Re: [OSL | CCIE_Voice] First Attempt...Failed miserably
Don't get down on yourself! Part of the CCIE pursuit is perseverance. Take some time today and try to visualize your experience yesterday, highs and lows. Think about what you could do better. Run through different scenarios in your head and then lab them out. Give me a call sometime (get cell through Twitter) if you want to chat. Thanks! Matthew Berry, CCIE #26721 (Voice) Email: thematthewbe...@gmail.com Twitter: http://twitter.com/CiscoVoiceGuru Tech Blog: http://ciscovoiceguru.com On 3/10/11 7:58 AM, adam compton wrote: 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] About Channel Selection Order (Julien Krieger)
Justin - When you take the lab, I *highly* suggest turning on debug isdn q931 and leave it on until you type the last wr. If you have that turned on, you'll be able to see what channel the PRI is sending inbound calls in on and then you can modify it accordingly. It's not that difficult of a thing to look out for. My recommendation would be to get in the habit of labbing with that debug turned on. That way, when you go into the lab it's already second nature. Matthew Berry Sr. Voice Engineer - CCIE 26721 http://www.cdw.com/content/services/advanced-technology/default.aspx CDW Advanced Technology Services 7145 Boone Avenue North | Brooklyn Park, MN 55428 Single Number Reach: +1.763.592.5987 matthew.ber...@cdw.com On 1/22/11 12:19 PM, Justin Barksdale jus...@barksdale.net wrote: It is best practice to set the channel selection order opposite what the PSTN provider does as to avoid glare. The lab is not so much best practice as it is doing what the proctor is asking in the manner in which they are asking. Justin Sent from my iPhone 4. On Jan 22, 2011, at 12:00 PM, ccie_voice-requ...@onlinestudylist.com wrote: Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: About Channel Selection Order (Julien Krieger) 2. Re: AAR display (Miron Kobelski) -- Message: 1 Date: Sat, 22 Jan 2011 17:11:42 +0100 From: Julien Krieger julien.krie...@ineo-gdfsuez.com To: bruno bruno.juni...@gmail.com Cc: ccie_voice ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] About Channel Selection Order Message-ID: a647ac29-4c68-49cc-8f1a-37d7d032c...@ineo-gdfsuez.com Content-Type: text/plain; charset=utf-8; Format=flowed; DelSp=yes Vick, guys, Can you please give us your thoughs on this. But again, unless clearly stated in your lab exam to do it one way, otherwise best practise applies. This is not something I made up. I am refering to IPExpert materials (books, videos) and SRND. Julien Le 22 janv. 2011 ? 13:07, bruno bruno.juni...@gmail.com a ?crit : hello, reverse or same chanel? why julien said it's same.Below is what he said You need to make sure that you are following the best practices in terms of what the SRND states with regards to GLARE. if your provider is sending you calls on the 1st channel (1 - 24), then you must send outgoing calls in the same selection order (1 - 24) but, if your provider is sending you calls on the 24th channel? (24 - 1), then you must send outgoing calls in the same selection order (24 - 1) Before configuring your side, make a call from a pstn source and find out the selection order of your provider. This is real life and CCIE lab -- Original -- From: George Goglidzegogli...@gmail.com; Date: Sat, Jan 22, 2011 02:54 AM To: brunobruno.juni...@gmail.com; Cc: ccie_voiceccie_voice@onlinestudylist.com; Subject: Re: [OSL | CCIE_Voice] About Channel Selection Order Hi Bruno, Just make sure it's reverse of what the Telco is using, to make sure you don't try to seize the same channel from both sides at the same time. Regards, 2011/1/21 bruno bruno.juni...@gmail.com Hello expert, ? I see this The Channel Selection Order?which defines the order the system ???hunts??? for available channels. The logic here is reverse d from a numerical perspective, as the channels are considered to be top at 1 and bottom at 24, therefore Top Down selection will select channel 1 first? in some doc.? i want to know what 's best pratice in real life or in ccie lab. top-down or bottom-up? ? Best Regards, Bruno ___ 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 -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20110122/c56023a6/attachment-0001.html -- Message: 2 Date: Sat, 22 Jan 2011 17:51:15 +0100 From: Miron Kobelski findko...@gmail.com To: Michael Luo hout...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com, Roger K?llberg roger.kallb...@cygate.se Subject: Re: [OSL | CCIE_Voice] AAR display Message-ID: AANLkTim+W-0U5=rcf3u7ipzr
Re: [OSL | CCIE_Voice] I passed my Voice CCIE
Way to go, Akash! Enjoy the afterglow and get some rest. After you come back from your hiatus, hit us up and tell us how life is with that number! Thanks, Matthew Berry, CCIE #26721 Senior Unified Communications Engineer CDW Single Number Reach: +1.763.592.5987 Email: matthew.be...@cdw.commailto:matthew.be...@cdw.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of akash patel Sent: Thursday, January 20, 2011 11:46 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] I passed my Voice CCIE I took my exam in San Jose and just found that I passed it, # 27992 I like to thank you Vik, Amy and entire IPExpert support team as well as everyone in this forum for outstanding help throughout my CCIE journey. I hope to stay active in this forum to help anyone with anything I can. Thank you all again, Akash ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] why trunk mode on ESW?
Just make sure you understand the differences and can configure either method, regardless of the equipment setup. Thanks, Matthew Berry Sr. Voice Engineer - CCIE 26721 Single Number Reach: +1.763.592.5987 matthew.ber...@cdw.commailto:matthew.ber...@cdw.com Check out my latest Guru Guide for QoS - http://ciscovoiceguru.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Rogers Ochieng Sent: Wednesday, January 19, 2011 7:27 AM To: Shrini Cc: ccie_voice; Roger Källberg; bruno Subject: Re: [OSL | CCIE_Voice] why trunk mode on ESW? All recommendations i see prefers the new way, even though out of our scope it's more secure as compared to trunk mode. On 19 January 2011 14:12, Shrini linuxbos...@gmail.commailto:linuxbos...@gmail.com wrote: Hi Roger, I agree with legacy word, but I prefer trunk for our purpose, reason is below link. access mode and trunk mode both explained well here. http://www.cisco.com/en/US/docs/ios/lanswitch/configuration/guide/lsw_hwic_ethsw_ic_ps6441_TSD_Products_Configuration_Guide_Chapter.html#wp1049866 -S From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roger Källberg Sent: Wednesday, January 19, 2011 2:13 AM To: bruno; ccie_voice Subject: Re: [OSL | CCIE_Voice] why trunk mode on ESW? Hi Bruno, First of all I do not speak for IPX, but my understanding is that the reason for why the vol1 has the old way of configuring the ports on ESW module is because of at the time of when it was written this was the way these ports were configured. I can from my own experience say that you definitely can configure these ports with the access port mode. This is what I did after I realized that this was now supported. But it might be good to practise on both methods, you never know if you will get that as a requirement in the lab. Sincerely Roger Källberg CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Från: bruno [bruno.juni...@gmail.commailto:bruno.juni...@gmail.com] Skickat: den 19 januari 2011 07:18 Till: ccie_voice Ämne: [OSL | CCIE_Voice] why trunk mode on ESW? Dear all in vol1 network infrastrure, why we need to configure trunk mode on esw,why not access mode .i have test the access mode is ok. SITEB(config)#int range f0/1/0 -3 SITEB(config-if-range)# switchport trunk native vlan 602 SITEB(config-if-range)# switchport mode trunk SITEB(config-if-range)# switchport voice vlan 502 SITEB(config-if-range)#description ***CONNECT TO IP PHONE*** SITEC#show interfaces f0/1/0 switchport Name: Fa0/1/0 Switchport: Enabled Administrative Mode: trunk Operational Mode: trunk Administrative Trunking Encapsulation: dot1q Operational Trunking Encapsulation: dot1q Negotiation of Trunking: Disabled Access Mode VLAN: 0 ((Inactive)) Trunking Native Mode VLAN: 602 (DATA-VLAN) Trunking VLANs Enabled: ALL Trunking VLANs Active: 1,502,602 Protected: false Priority for untagged frames: 0 Override vlan tag priority: FALSE Voice VLAN: 502 Appliance trust: none Best Regards, bruno ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Cisco 79XX remote control
Singlewire has a Remote Phone Control software that is good for large sites. You might also check out VoIP Integrations. They have a great app for controlling phones. Probably much cheaper than the Singlewire one. The same company has a handy tool for pushing out new background images to phones. Matthew Berry Sr. Voice Engineer - CCIE 26721 [cid:11BC3241-56E2-4B23-BF63-ABEDB5FF9BBC]http://www.cdw.com/content/services/advanced-technology/default.aspx CDW Advanced Technology Services 7145 Boone Avenue North | Brooklyn Park, MN 55428 Single Number Reach: +1.763.592.5987 matthew.ber...@cdw.com From: haroon javed harooon.ja...@gmail.commailto:harooon.ja...@gmail.com Date: Tue, 18 Jan 2011 01:44:49 -0600 To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Cisco 79XX remote control Does anyone know of any preferably software that will allow you to mange the Cisco 79xx series of phones remotely. We have a large amount of phones that are staticly assinged and are moving to a new CM server with a different IP. The CM and TFTP entries specifcly need to be change to reflect the new IP of the CM server. Thanks -- Regards, Haroon Javed Telecom Engineer Cell: +92 (321) 8430260 inline: F906EF75-D025-431C-B55C-27FF496CF05D[1].png___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] New Blog Post: CCIE Voice Strategy
Hey everyone. I posted an new article on my blog. I thought it would apply to most of you out there. I'm interested to get your feedback on it as part of the CCIE Voice community. http://matthewberry.info/ciscovoiceguru/548/ccie-voice-lab-strategy/ Here's the first paragraph: When approaching the CCIE Voice lab exam it’s important to have a well-honed strategy that you have relentlessly practiced time and time again. This isn’t your typical CCNA exam that you can cram for the night before, walk in, and pass with flying colors. The CCIE lab is a sacred gauntlet that will test time management, reasoning under pressure, and technical prowess. By “prowess” I am not referring to whether you know how to configure and CUCME in SRST mode while preserving CUCM-based features. I am referring to the ability to take a question, consider the restrictions stated in previous questions, and configure a solution that is quirky and contrary to the standard examples you see in the DocCD…. Matthew Berry Sr. Voice Engineer - CCIE 26721 [cid:F404D420-1013-431B-977D-0F0EFEEAADD1]http://www.cdw.com/content/services/advanced-technology/default.aspx CDW Advanced Technology Services 7145 Boone Avenue North | Brooklyn Park, MN 55428 Single Number Reach: +1.763.592.5987 matthew.ber...@cdw.com inline: F906EF75-D025-431C-B55C-27FF496CF05D.png___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Back ground image on CME phones
Also, make sure you have the web service turn on the router as well as the path for web pages. Those, along with the list.xml and image files are needed to setup background images in CME. Thanks, Matthew Berry Sr. Voice Engineer - CCIE 26721 CDW Advanced Technology Services 7145 Boone Avenue North | Brooklyn Park, MN 55428 Single Number Reach: +1.763.592.5987 matthew.ber...@cdw.commailto:matthew.ber...@cdw.com On Jan 11, 2011, at 4:49 AM, Friderich Claude wrote: Rahul, Did you copy the List.xml form phone administration guide ?? Why this question ? :) Because I had the following problem. CiscoIPPhoneImageList ImageItem Image=TFTP:Desktops/320x212x16/xx-tn.png URL=TFTP:Desktops/320x212x16/xxx.png/ /CiscoIPPhoneImageList I put the double quotes manually again because copy paste to notepad was bad for me ….. Perhaps you have the same pb ….. Regards Claude. Claude Friderich PreSales Support image001.gif NETCORE PSF S.A. 49 rue du Baerendall B.P.65 L-8201 Mamer Téléphone: 31 33 80-407 Fax: 31 33 80 8-407 GSM: 621 303 616 E-mail: cfrider...@netcore.lumailto:cfrider...@netcore.lu From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf OfRahul Kapor Sent: mardi 11 janvier 2011 10:40 To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Back ground image on CME phones Hi Mates, I tried to upload the image on 7965 phones registered to CME. Respective files are uploaded to tftp server tftp-server flash:Desktops/320x212x16/TNImage.png tftp-server flash:Desktops/320x212x16/MAINImage.png tftp-server flash:Desktops/320x212x16/List.xml did create cnf file under telephony service and reset the phones several times but no luck :( same file are upload to cucm and working fine. Any clue here ? thx, Rahul -- This email was Anti Virus checked. ATT1..txt ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] GREETINGS...
Thanks for the mention, Steven. Welcome to the community! If you have any questions I'd be happy to help. I'm on a big project right now, but if I can spare the time it's always great to help candidates in their pursuit. Stay focused! Get sleep! Make a study plan! Ready, set, go! Thanks, Matthew Berry Sr. Voice Engineer - CCIE 26721 Single Number Reach: +1.763.592.5987 matthew.ber...@cdw.commailto:matthew.ber...@cdw.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Steven Juras Sent: Sunday, January 09, 2011 7:34 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] GREETINGS... Hello all - I've been following the Voice-OSL now for many months as I've been passively studying for the Voice Lab since successfully passing my written-exam back in March 2010. I've been sort of sitting on the sidelines up till now as I'm now officially hitting the road with focused studies! I've slated my lab sitting for May 10th at RTP and so I'm excited to be coming on the field so to speak. I've started a blog at www.jurassiclabs.nethttp://www.jurassiclabs.net/ and I invite you to take a look over the next several weeks as I will try to make relevant posts about my studies / progress. I've learned a lot of pointers from some of you (Matthew Berry) and appreciate your posts as both motivation and guidance! Thanks to you guys! You can also follow me on Twitter at @stevenjuras . I'm not a big social media nut so you won't see me post time wasting info about what I ordered for lunch. It's my desire and goal to put these new tools to professional use and join the community. Steven Juras - CCNP, CCDP, CCVP www.jurassiclabs.nethttp://www.jurassiclabs.net/ @stevenjuras Thomasville, Georgia ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Inbound SIP call to CUCM from CUBE goes unanswered
You could do a debug ccsip messages on the CUBE and see what's taking place in the SIP messages between the gateway and CUCM. Send that on over for us to take a look at. Matthew Berry Sr. Voice Engineer - CCIE 26721 [cid:CB259CF6-BDEC-4D6C-AE3F-AC1ADFE83021]http://www.cdw.com/content/services/advanced-technology/default.aspx CDW Advanced Technology Services 7145 Boone Avenue North | Brooklyn Park, MN 55428 Single Number Reach: +1.763.592.5987 matthew.ber...@cdw.com From: ccielab...@gmail.commailto:ccielab...@gmail.com ccielab...@gmail.commailto:ccielab...@gmail.com Date: Fri, 31 Dec 2010 08:59:20 -0600 To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Inbound SIP call to CUCM from CUBE goes unanswered I'm testing a cube configuration in my lab setup. I have H.323 coming from CME to CUBE running on R1 and then SIP to the CUCM via a SIP trunk. I see the proper dialpeers being triggered in CUBE, but the CUCM doesn't seem to respond to the SIP call setup inbound. Calls from CUCM to CME via CUBE work , so I'm pretty confident the SIP trunk is functional. Short of trying to look through CUCM traces, is there a good debug on R1/Cube that would provide some insight into whats going on? inline: F906EF75-D025-431C-B55C-27FF496CF05D[6].png___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] # of channels to bring up on E1/T1
The following statement does not break NDA: When you take the lab, there will be no question how many channels you need. If anything, you'll understand that the exam questions are precisely worded. You have to learn to pick up nuances within the questions. However, that said, there's pretty much only one way to say 10 channels or all channels. :) Matthew Berry Sr. Voice Engineer - CCIE 26721 [cid:62A9CDAA-71BB-40C8-B7AC-EFB9C484EB25]http://www.cdw.com/content/services/advanced-technology/default.aspx CDW Advanced Technology Services 7145 Boone Avenue North | Brooklyn Park, MN 55428 Single Number Reach: +1.763.592.5987 matthew.ber...@cdw.com From: Julien Krieger krieger.jul...@gmail.commailto:krieger.jul...@gmail.com Date: Tue, 28 Dec 2010 12:30:39 -0600 To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] # of channels to bring up on E1/T1 Hi all, I wonder how many channels are we supposed to bring up when setting up a E1/T1 controller when the question does not specify anything about it? Not having any clue about the number of channel configured on the PSTN side, how can we know how many channels to bring up? Thank you, Julien inline: F906EF75-D025-431C-B55C-27FF496CF05D[29].png___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] call from CUCM to CME : VIA Zone
Correct, you do not need to register your CUBE with a tech prefix at all. Matthew Berry Sr. Voice Engineer - CCIE 26721 [cid:E1F2A7B1-98EB-4A5E-861A-C1BB303B86B6]http://www.cdw.com/content/services/advanced-technology/default.aspx CDW Advanced Technology Services 7145 Boone Avenue North | Brooklyn Park, MN 55428 Single Number Reach: +1.763.592.5987 matthew.be...@cdw.commailto:ben.c...@cdw.com From: givemeccievoice2...@gmail.commailto:givemeccievoice2...@gmail.com givemeccievoice2...@gmail.commailto:givemeccievoice2...@gmail.com Date: Tue, 30 Nov 2010 07:49:07 -0600 To: 'Mritunjay Kumar' mjs...@gmail.commailto:mjs...@gmail.com, ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] call from CUCM to CME : VIA Zone Why do you have a different tech prefix for VIA zone? I don’t believe you need a tech prefix at all for a VIA zone / CUBE. Just have your dial-peers configured to receive what CUCM is sending. Also, make sure that you have your “allow connections” commands. Do a show gatekeeper endpoints and CUBE should be registered as “H323-GW”. If not, make sure those commands are present and/or bounce the gateway command. Please post the debug gatekeeper main 10 output as well as show gatekeeper end. Jeff From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mritunjay Kumar Sent: Tuesday, November 30, 2010 3:06 AM To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] call from CUCM to CME : VIA Zone Hi all , I am facing issue in GK and VIA zone sh gatek gw CUCM is registered to GK using tech-p 1# CME is registered to DK using tech-p 44# VIA zone is registered to GK using tech-p 12# sh gatek gw Prefix: 1#* Zone CUCM master gateway list: 14.160.110.15:1720http://14.160.110.15:1720 US_1 Zone CUBE master gateway list: 14.160.110.254:1720http://14.160.110.254:1720 MJ-CUBE Prefix: 12#* Zone CUBE master gateway list: 14.160.110.254:1720http://14.160.110.254:1720 MJ-CUBE Prefix: 44#* Zone CME master gateway list: 14.160.115.200:1720http://14.160.115.200:1720 MJ-CME call from CME to CUCM is working fine but call from CUCM to CME is failing. while calling this i am adding correct tech prefix and removing it in CME enabled debug gate main 5 and error is *Nov 30 11:01:04.435: //001191430300/001191430300/GK/rassrv_get_addrinfo: (44#5002) Matched tech-prefix 44# assrv_get_addrinfo(44#5002): Viazone gateway selection failed for zone CUBE . when CME is registered with tech-p 1# , everthing works fine ie call in both direction without introducing VIA zone , everyting works fine ie CME is regiserted with tech-p 44# gatekeeper cofig gatekeeper zone local BR1 cisco.comhttp://cisco.com 14.160.110.129 zone local CME cisco.comhttp://cisco.com invia CUBE outvia CUBE zone local CUCM cisco.comhttp://cisco.com invia CUBE outvia CUBE zone local CUBE cisco.comhttp://cisco.com zone prefix CUCM 2... zone prefix CME 5... gw-type-prefix 1#* no shutdown Is any configmissing ?? Regards, MJ inline: F906EF75-D025-431C-B55C-27FF496CF05D[3].png___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME- Can not call Vocie Hunt Group Pilot from PSTN Phone
Can you send your configs? It's hard to troubleshoot without anything to work with. Thanks! Matthew Berry On Nov 26, 2010, at 7:22 PM, vccie2010 wrote: Folks, CME- Can not call Vocie Hunt Group Pilot from PSTN Phone, get Fast busy and ISDN message unallocated/ unassingned number, I can call from another CME phone though. I see the call hitting the CME Thanks for your help AB ATT1..txt ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] WB1 Lab 3A
The easiest way is to home that phone to CUCM, have CUCM set for SCCP auto-registration, let the firmware updated, then swing it back to CUCME. Matthew Berry Sr. Voice Engineer - CCIE 26721 [cid:DA4125F0-9AB2-4CB2-A16C-E5E2755B4790]http://www.cdw.com/content/services/advanced-technology/default.aspx CDW Advanced Technology Services 7145 Boone Avenue North | Brooklyn Park, MN 55428 Single Number Reach: +1.763.592.5987 matthew.be...@cdw.commailto:ben.c...@cdw.com From: Dew Swen dew.s...@gmail.commailto:dew.s...@gmail.com Date: Wed, 17 Nov 2010 10:15:14 -0600 To: CCIE Voice Maillist ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] WB1 Lab 3A Hey guys, I am in the middle of the session. BR2Ph1 has the SIP firmware loaded. I have to register it to BR2CME with SCCP firmware. How can change its firmware from SIP to SCCP. Because of it is a remote phone, I cannot change the TFTP IP address from the phone. Any comments? -- Dew Swen inline: F906EF75-D025-431C-B55C-27FF496CF05D[1].png___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Support Needed!
good luck I hope you do well in your lab! Thanks, Matthew Berry Sent from my Verizon Wireless Phone. Please excuse brevity and any typo errors. - Reply message - From: Scott Newberry sc...@meganandscott.com Date: Sun, Nov 14, 2010 3:26 pm Subject: [OSL | CCIE_Voice] Support Needed! To: osl osl ccie_voice@onlinestudylist.com Anybody know how to contact ProctorLabs After-Hours Support if you're not in a current rack session? I had sessions scheduled for today that are no longer scheduled. And of course, I can't schedule now since it's past the start time. My lab exam is tomorrow... Just wanted to run through a few things. Scott ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 10 WB 1 question --QoS
When I was using the remote racks back in the summer, there was a known issue when using auto QOS on their equipment due to IOS version mismatch on the Proctor Labs gear. Do a show version to check. Thanks, Matthew Berry Sent from my Verizon Wireless Phone. Please excuse brevity and any typo errors. - Reply message - From: Rrcrumm rrcr...@yahoo.com Date: Sun, Nov 14, 2010 9:06 pm Subject: [OSL | CCIE_Voice] Lab 10 WB 1 question --QoS To: Randall Crumm rrcr...@yahoo.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Hi I run the auto QoS command on the hq router fine for the interface towards br2. When I run auto QoS for the interface for br1 it runs but does not create the policy map or class map. Has anyone seen this issue? Thanks Randall Sent from my iPhone On Nov 14, 2010, at 3:47 PM, Randall Crumm rrcr...@yahoo.commailto:rrcr...@yahoo.com wrote: HI I am working on lab 10 wb 1 I am getting this error on the HQ router when applying auto qos to the DLCI for br1 Nov 15 04:41:49.345: %RMON-5-FALLINGTRAP: Falling trap is generated because the value of cbQosCMDropBitRate.354.12774241 has fallen below the falling-threshold value 0 Any ideas? Thanks in advance. Randall ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.com www.ipexpert.comhttp://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Checking In
Hey guys. It's funny how once you pass comes this realization that the world actually doesn't revolve around the CCIE certification process. I started a new job at CDW which has kept me extremely busy. Was it worth it? Absolutely. I know what I'm capable of. Passing the CCIE is a major accomplishment, especially the Voice IE. Of course, with the IE comes more money, better job/position, etc. (To answer your question). Am I going to start ciscoROUTINGguru.com? Not any time soon! I'm ready to get back to life as normal. Maybe pick up a few specialist certs along the way. ;) Maybe one day I'll get into my blog again. For now, I need to take a hiatus. :) I'm happy to answer any non-NDA questions you guys may have. I was deeply indebted to those folks who mentored me along the way and I'd like to repay the favor however I can. Matthew Berry Sr. Voice Engineer - CCIE 26721 [cid:DD43E6B6-4DEE-4FA7-AD30-3033A25D58F1]http://www.cdw.com/content/services/advanced-technology/default.aspx CDW Advanced Technology Services 7145 Boone Avenue North | Brooklyn Park, MN 55428 Single Number Reach: +1.763.592.5987 matthew.be...@cdw.commailto:ben.c...@cdw.com From: Paul Kruger pauld.kru...@gmail.commailto:pauld.kru...@gmail.com Date: Sat, 13 Nov 2010 07:53:21 -0600 To: Matthew Berry matthew.be...@cdw.commailto:matthew.be...@cdw.com Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Checking In Welcome back Legendary One! How's life with a number? Relaxing more due to not studying, or more stress due to increased (advanced) work demands? Major job/position/salary changes after passing, or business as usual? Good to see you popping in! We (I) miss your regular blog updates. but I understand. Take care, Paul On Sat, Nov 13, 2010 at 3:31 AM, Matthew Berry matthew.be...@cdw.commailto:matthew.be...@cdw.com wrote: Hi everyone. I thought I'd join the OSL again after being away for the past few months. Thought I'd say hi. Matthew Berry Sr. Voice Engineer - CCIE 26721 [cid:320BCC04-AF56-4810-AA83-9B4831FDCB1D]http://www.cdw.com/content/services/advanced-technology/default.aspx CDW Advanced Technology Services 7145 Boone Avenue North | Brooklyn Park, MN 55428 Single Number Reach: +1.763.592.5987 matthew.be...@cdw.commailto:ben.c...@cdw.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com inline: F906EF75-D025-431C-B55C-27FF496CF05D[47].pnginline: F906EF75-D025-431C-B55C-27FF496CF05D[49].png___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Checking In
Hi everyone. I thought I'd join the OSL again after being away for the past few months. Thought I'd say hi. Matthew Berry Sr. Voice Engineer - CCIE 26721 [cid:320BCC04-AF56-4810-AA83-9B4831FDCB1D]http://www.cdw.com/content/services/advanced-technology/default.aspx CDW Advanced Technology Services 7145 Boone Avenue North | Brooklyn Park, MN 55428 Single Number Reach: +1.763.592.5987 matthew.be...@cdw.commailto:ben.c...@cdw.com inline: F906EF75-D025-431C-B55C-27FF496CF05D[47].png___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] GK Calls from gateway failing
If you're dealing with 1001 and 3001 extensions, why is the gatekeeper handling 1#5001? It looks like you're translating the number incorrectly on BR2. Please send the following: - Copy of gatekeeper configuration - Dial-peers on both gateways - debug ras output on initiating gateway Matthew Berry Sr. Voice Engineer - CCIE 26721 [cid:917D4399-502C-4878-B05A-75EF8058FF71]http://www.cdw.com/content/services/advanced-technology/default.aspx CDW Advanced Technology Services 7145 Boone Avenue North | Brooklyn Park, MN 55428 Single Number Reach: +1.763.592.5987 matthew.be...@cdw.commailto:ben.c...@cdw.com From: Shrini linuxbos...@gmail.commailto:linuxbos...@gmail.com Date: Fri, 12 Nov 2010 21:04:28 -0600 To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] GK Calls from gateway failing Hi, I have HQ site configured as GK and Br2 as GK controlled. When I call from 1001 (HQ DN) to 3001 (Br2 DN) call goes fine, but reverse is not working. Here is the log: R1#debug gatekeeper main 10 R1# *Nov 13 03:04:05.055: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup *Nov 13 03:04:05.055: ////GK/gk_rassrv_arq: arqp=0x4A387188,crv=0x1E, answerCall=0 *Nov 13 03:04:05.059: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC *Nov 13 03:04:05.059: //6C5631E3809A/6C5631E3809C/GK/gk_dns_query: No Name servers *Nov 13 03:04:05.059: //6C5631E3809A/6C5631E3809C/GK/rassrv_get_addrinfo: (1#5001) Matched tech-prefix 1# *Nov 13 03:04:05.059: //6C5631E3809A/6C5631E3809C/GK/rassrv_get_addrinfo: (1#5001) Matched zone prefix 1 and remainder 001 *Nov 13 03:04:05.059: ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 1 *Nov 13 03:04:05.059: //6C5631E3809A/6C5631E3809C/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x49B6AB2C *Nov 13 03:04:05.059: //6C5631E3809A/6C5631E3809C/GK/rassrv_arq_select_viazone: matched zone is ZONE_01, and z_invianamelen=0 * R1#Nov 13 03:04:05.059: //6C5631E3809A/6C5631E3809C/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x49B6AB2C *Nov 13 03:04:05.059: //6C5631E3809A/6C5631E3809C/GK/rassrv_arq_select_viazone: matched zone is ZONE_01, and z_outvianamelen=0 *Nov 13 03:04:05.059: ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 1 R1# *Nov 13 03:04:08.855: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup *Nov 13 03:04:09.719: ////GK/gk_process: got a TIMER event *Nov 13 03:04:09.719: ////GK/gk_handle_timers *Nov 13 03:04:09.719: ////GK/gk_handle_timers: managed timer expired 0x471677E0 inline: F906EF75-D025-431C-B55C-27FF496CF05D[48].png___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Post-Lab Reflections and Thanks
All - I want to thank IPexpert for their help and support during the last year. Amy and Vik are incredible instructors. They really want to form you into a true CCIE, not just someone who can pass the test. I am greatly indebted to the amount of work they put into the v3 materials that they offer. Way to go guys! I also want to say a HUGE thanks to the IPexpert technical support guys: Drew LePla, Ryan Barnum, and Andrew B Shipton. You guys know how many times I sent after-hours support requests! Thank you for your help. A thanks to my buddy, Mike Down aka Frank - You sold me a good deal on the end-to-end package and provided plenty of sarcasm and customer service throughout my journey. Keep yer' stick on the ice, my friend. I also want to thank my study partners: Antonio McCarver, Roger Källberg, Jeff Cotter, Warren Heaviside, and the list goes on and on. I made some great friends on this journey. You know who you are. Let's keep in touch. I also wanted to shoot out a few thoughts while things were still fresh in my mind. Tip 1: You begin taking the lab the night before Make sure that you prepare yourself for the lab the night before. My wife told me to not have sugar or carbs because they can slow down your mental recall abilities. Don't eat heavy food. Try to avoid excess sugar and carbs. Take a 30-45 minute walk the night before. This will help alleviate stress and provide feel-good endorphins that will help as you go to bed. The morning of the exam, do not have ANY sugar or carbs. For me, I went to Denny's and had eggs, bacon, and fruit. Protein is good for endurance and mental alertness. After having breakfast, I went for a 30 minute walk. I was super nervous going into the lab because it was my first attempt. I felt that the walk in the morning was a great stress reliever. When I went into the lab, I was riding high on those positive endorphins for the first hour. Tip 2: Don't waste your free meal at Cisco's cafeteria You get something like $13-14 to spend for lunch. Following my wife's advice, I avoided sugar and carbs. I had a big salad with tons of protein (chicken, bacon, eggs) and fruit. I was tempted by the fresh pizza, burgers, and fries, but managed to avoid them. When I returned to the lab, I was alert and not groggy in any way. Other guys picked up sugary drinks, chocolate, cookies, fries, etc. Don't make that mistake! You've invested a lot of time into your preparation, don't handicap yourself by being undisciplined and eating junk food for lunch. Tip 3: Keep a spreadsheet to track your study progress The CCIE lab requires a high level of personal dedication and perseverance. Use a spreadsheet to track your study time. Every Monday morning, I would determine the number of hours I would study that week, clearly define what IPexpert labs I would focus on, and what Cisco documents/concepts I would study. I would schedule my week and hold myself to it. Logging your time can be a great confidence-booster as well. By the time I went in to the take the test, I had logged 600 hours worth of rack time and another 350 hours worth of reading/reflection since January 1st. I was able to confidently tell myself, Matthew, you know this! You've done this many, many times before. Tip 4: Get involved in online study lists like the OSL Gather people around you who will challenge you. IPexpert's online study list was a great way to meet other people and be challenged. If you come across a question, do your research, check the OSL archives, and then send an email out to the group if you're still stumped. Make an effort to be a contributor. Don't just ask questions, but answer them as well. I made a commitment early on to answer at least one email once a week. It was a great way to be stretched. I will write more on my blog over the next few weeks, but these were just a few tips that really helped me. You know, there's no shortcut to getting your CCIE. In the end, it takes a lot of hard work, sore muscles, awkward schedules, etc. The reason Voice IEs are so coveted in the marketplace is because there are so few of them. Not many people are willing to make the sacrifice in order to get the prize. Commit yourself to the goal, throw yourself into your study plan, and get 'er done! Thanks, Matthew Berry ciscovoiceg...@gmail.com http://ciscovoiceguru.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CCIE #26721 - I PASSED!
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
Re: [OSL | CCIE_Voice] Gatekeeper IPIPGW
Good description ! Matthew Berry **Sent from my iPhone** Skype/Twitter: ciscovoiceguru Google Voice: +1 612 424 5044 On Aug 13, 2010, at 16:30, Matthew Hall 1.matt.h...@gmail.com wrote: outvia sends the call to the zone specified on the way to the destination zone (like a next hop statement in a policy routebut not really). The gatekeeper then just looks for a cube registered in that outvia zone. The cube has to have a matching inbound voip dialpeer AND a outgoing dialpeer with a destination pattern that matches the outbound digits and session target RAS. The CUBE uses that dial-peer to send the call back to the gatekeeper, the gatekeeper knows then to send the call on to the final zone without looping back to the CUBE. Matt On Aug 12, 2010, at 1:19 PM, CCIE Voice GMAIL wrote: Hi Matt, I wanted to see what the logic was behind the zone local statement. Is it good practice to do it that way (invia and outvia) for Intra-zone GK routing? From what I would understand, this is almost implied by just saying zone local GK Can you share with me your reasoning for doing it this way? Thanks in advance. -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccieid1ot Sent: Thursday, August 12, 2010 8:40 AM To: Matthew Berry Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Gatekeeper IPIPGW Use this on your local GK. gatekeeper zone local GK cisco.com x.x.x.x invia GK outvia GK enable-intrazone zone remote BBGK cisco.com x.x.x.x1719 zone prefix BBGK 01132* zone prefix BBGK 01144* no shutdown On Wed, Aug 11, 2010 at 9:01 PM, Matthew Berry ciscovoiceg...@gmail.com wrote: Edwin - You need to add the outvia command to the end of your remote zone and specify the zone that has your IPIPGW. Thanks, Matthew Berry ciscovoiceg...@gmail.com http://ciscovoiceguru.com On Aug 11, 2010, at 7:13 PM, Edwin Dotson wrote: Sorry for all the same topic. I have successfully got my 2 gatekeepers talking and calls back and fourth. But now I can’t get calls to pass through the IPIPGW it sends the calls directly to the Remote Gatekeeper. To identify calls destined for the IPIPGW I have been using 3#. LOCAL GK/IPIPGW version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname LOCAL ! boot-start-marker boot system flash flash:c3825-adventerprisek9_ivs-mz.124-25c.bin boot-end-marker ! enable password ! no aaa new-model ip cef ! ! ! ! ip auth-proxy max-nodata-conns 3 ip admission max-nodata-conns 3 ! voice-card 0 no dspfarm ! ! ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip supplementary-service h450.12 ! ! ! ! ! ! ! ! ! ! voice translation-rule 1 rule 1 /^3#/ // ! ! voice translation-profile strip3# translate called 1 ! ! ! ! ! ! ! ! ! ! ! interface GigabitEthernet0/0 ip address 10.201.3.30 255.255.255.0 duplex auto speed auto media-type rj45 ! interface GigabitEthernet0/1 ip address 172.24.200.5 255.255.255.0 duplex auto speed auto media-type rj45 h323-gateway voip interface h323-gateway voip id GK ipaddr 172.24.200.5 1719 h323-gateway voip h323-id CUBE h323-gateway voip tech-prefix 3# h323-gateway voip bind srcaddr 172.24.200.5 ! ip forward-protocol nd ! ! ip http server no ip http secure-server ! ! ! ! control-plane ! ! ! dial-peer voice 12 voip description Incoming Dialplan translation-profile incoming strip3# session target ras incoming-called number . dtmf-relay h245-alphanumeric! ! ! gateway timer receive-rtp 1200 ! ! ! ! gatekeeper zone local GK 172.24.200.5 zone remote REMOTEGK cisco.com 172.24.200.6 1719 zone prefix GK 3* zone prefix REMOTEGK 5... zone prefix GK 6... gw-type-prefix 1#* default-technology no shutdown ! ! line con 0 line aux 0 line vty 0 4 password login ! scheduler allocate 2 1000 ! end Remote Gatekeeper/CME version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname REMOTE ! boot-start-marker boot system flash flash:c3845-adventerprisek9_ivs-mz.124-25c.bin boot-end-marker ! enable password ! no aaa new-model voice-card 0 no dspfarm ! voice-card 1 no dspfarm ! ip cef ! ! ! ! ip auth-proxy max-nodata-conns 3 ip admission max-nodata-conns 3 ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! archive log config hidekeys ! ! ! ! ! ! ! interface GigabitEthernet0/0 ip address 172.24.200.6 255.255.255.0 duplex auto speed auto media-type rj45 h323-gateway voip interface h323-gateway voip id REMOTEGK ipaddr 172.24.200.6 1719 h323-gateway voip h323-id CME h323-gateway voip tech-prefix 1# h323-gateway voip bind srcaddr 172.24.200.6
[OSL | CCIE_Voice] Calling Number Display on Phone - Calls to PSTN-WAN
All - I'm sending calls to 9011.91!# across the local gatekeeper to the PSTN-WAN backbone gatekeeper. What I notice is that no matter what I try to do in order to manipulate the calling number displayed on the IP phone, it always shows 6745738932. Meaning, if I dial 9011-91-67-4573-8932 on my Cisco IP phone, the display shows up as 6745738932. I am telling the system to stripp the 9011, but where does the 91 go? I was told by a friend that this is due to the num-exp command used on the PSTN-WAN backbone gatekeeper. Is that true? I'm concerned that this may bite me in the lab. The no supplementary-service h225-notify cid-update command does not fix this. Ideas? Thanks, Matthew Berry ciscovoiceg...@gmail.com http://ciscovoiceguru.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] VPIM License Missing?
Is anyone aware of a VPIM license issue with ProctorLabs? When I try to add VPIM locations on CUC, I get a license error. inline: Shot 2010-08-09 at 10.13.21 PM.png Thanks, Matthew Berry ciscovoiceg...@gmail.com http://ciscovoiceguru.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] MGCP Fallback in SRST
Guys - I need some clarification on MGCP configuration for SRST. Looking through Cisco documentation, I see the following snippet from time to time: application global service alternate Default In my studies the past year, I have never used those commands and my testing has always worked as expected. Typically, I will configure the following: mgcp mgcp call-agent 10.10.210.11 service-type mgcp version 0.1 mgcp dtmf-relay voip codec all mode out-of-band no mgcp timer receive-rtcp mgcp bind control source-interface Loopback0 mgcp bind media source-interface Loopback0 ! ccm-manager switchback immediate ccm-manager fallback-mgcp ccm-manager redundant-host 10.10.210.10 ccm-manager mgcp ccm-manager music-on-hold HQ-RTR#show ccm-manager fallback-mgcp Current active Call Manager:10.10.210.11 MGCP Fallback mode: Enabled/OFF Last MGCP Fallback start time: None Last MGCP Fallback end time:None Can anyone shed light on the commands I mentioned at the beginning of this email? I understand that the command is supposed to let MGCP transition into H.323 and terminate the L3 backhaul to CUCM. But does it actually work and is it actually required? Thanks, Matthew Berry ciscovoiceg...@gmail.com http://ciscovoiceguru.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] VPIM between CUE and CUC
Joaquim - Check out my blog at ciscovoiceguru.com. I just recently did a post on all the CUE documentation through the CIsco website. On Aug 6, 2010, at 9:50 AM, Joaquim Fernandes wrote: HI TAPAN, Thanks for your reply. i know about this link. but i need for CUC and in pushkar it was unity. Regards, Jf --- On Fri, 8/6/10, Tapan Gautam (tgautam) tgau...@cisco.com wrote: From: Tapan Gautam (tgautam) tgau...@cisco.com Subject: RE: [OSL | CCIE_Voice] VPIM between CUE and CUC To: Joaquim Fernandes joa_...@yahoo.com, ccie_voice@onlinestudylist.com Date: Friday, August 6, 2010, 7:40 PM Try this http://pushkarbhatkoti.wordpress.com/category/cue-voicemail-vpim-networking-cue-to-unity-in-10-minutes/ From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Joaquim Fernandes Sent: Friday, August 06, 2010 8:23 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] VPIM between CUE and CUC Hi Team, Is there any document where i can configure VPIM between CUE and CUC. I am hunting over the google but havent found anything great. I need a document giving step by step procedure for configuring vpim between CUC AND CUE. Thanx in advance. Regards, JF ___ 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] Location of QoS SRND from DocCD Page
I know the QoS SRND is available through cisco.com/go/design, but I am trying to find it from the http://www.cisco.com/cisco/web/psa/default.html web page. Does anyone know the location of the document when accessed from the link above? Thanks, Matthew Berry ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Great News: Another CCIE for Rick Mur!
Way to go! Next stop, Voice! On Aug 4, 2010, at 9:34 PM, Marko Milivojevic wrote: Hello everyone, Sorry for cross post, but I just wanted to share with you that one of our Senior Support Engineers, Rick Mur, passed his Storage lab in San Jose yesterday. This is Rick's 3rd CCIE and to make things more fun - he passed ALL THREE on first attempt! -- Marko Milivojevic - CCIE #18427 Senior Technical Instructor - IPexpert FREE CCIE Training: http://bit.ly/vLecture Mailto: mar...@ipexpert.com Telephone: +1-810-326-1444 Web: http://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] Layer 2 QOS
You could treat it as a priority queue by throwing CoS 5 or DSCP EF into Q1. You could then shape it to 10, which would result in 10%. You would also need to do a no priority-queue out under the interface. But I don't think you can have priority on the queue and still limit the queue to only part of the pipe. Matthew Berry **Sent from my iPhone** Skype/Twitter: ciscovoiceguru Google Voice: +1 612 424 5044 On Jul 29, 2010, at 20:19, Jeff Cotter jcot...@voxns.com wrote: How would you enable the priority queue AND make sure queue 1 has 10% of the bandwidth. The documentation states that if the priority queue in enabled, shape and share configuration for that queue is ignored. So how do you accomplish this without using Shape command. ___ 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] Layer 2 QOS
Jeff - I've struggled with the same issue. This is something I'd really like Vik or Amy to comment on. If we need to treat traffic as priority while also dedicating 10% to that traffic, how would you do that? I thought Vik said to just remove the priority-queue out command under the interface (during our OWLE) but it still doesn't seem right to me. Any others? Matthew Berry **Sent from my iPhone** Skype/Twitter: ciscovoiceguru Google Voice: +1 612 424 5044 On Jul 29, 2010, at 20:56, Jeff Cotter jcot...@voxns.com wrote: Hello Matthew and thanks for the reply. However my thought is……putting COS 5 and EF into Q1 does not make it a priority queue. Which by definition means the queue is serviced until it is empty BEFORE the other queues are serviced. This behavior is only in effect with the priority queue out command. The Ingress queues have the proper commands to control the size of the priority queue mls qos srr-queue input priority-queue queue-id bandwidth weight but not the egress queues. Of course I my logic could be flawed here. From: Matthew Berry [mailto:ciscovoiceg...@gmail.com] Sent: Thursday, July 29, 2010 6:48 PM To: Jeff Cotter Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Layer 2 QOS You could treat it as a priority queue by throwing CoS 5 or DSCP EF into Q1. You could then shape it to 10, which would result in 10%. You would also need to do a no priority-queue out under the interface. But I don't think you can have priority on the queue and still limit the queue to only part of the pipe. Matthew Berry **Sent from my iPhone** Skype/Twitter: ciscovoiceguru Google Voice: +1 612 424 5044 On Jul 29, 2010, at 20:19, Jeff Cotter jcot...@voxns.com wrote: How would you enable the priority queue AND make sure queue 1 has 10% of the bandwidth. The documentation states that if the priority queue in enabled, shape and share configuration for that queue is ignored. So how do you accomplish this without using Shape command. ___ 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] Layer 2 QOS
So what's the answer? Vik/Amy - Opinions? Matthew Berry **Sent from my iPhone** Skype/Twitter: ciscovoiceguru Google Voice: +1 612 424 5044 On Jul 29, 2010, at 21:21, Jeff Cotter jcot...@voxns.com wrote: Interesting……Thanks Daniel great thought! From: Daniel Berlinski [mailto:dberlin...@gmail.com] Sent: Thursday, July 29, 2010 7:03 PM To: Jeff Cotter Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Layer 2 QOS In my opinion this is done by adjusting the buffer size for queue 1 and applying it to a queue-set. srr shape statement in my opinion means nothing in relation to adjusting priority queue size. http://onlinestudylist.com/archives/ccie_voice/2010-July/069398.html On Fri, Jul 30, 2010 at 1:19 PM, Jeff Cotter jcot...@voxns.com wrote: How would you enable the priority queue AND make sure queue 1 has 10% of the bandwidth. The documentation states that if the priority queue in enabled, shape and share configuration for that queue is ignored. So how do you accomplish this without using Shape command. ___ 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] QoS question about uplink port
Guys - When configuring QoS on an uplink port, how do I determine whether to trust CoS or DSCP markings? I always thought that you would trust CoS markings on access ports with IP phones on the other end since the phone will mark packets as CoS3 (signaling) or CoS 5 (media). The access ports connected to servers would be configured to trust DSCP since CUCM marks according to DSCP. My understanding is that the mls qos trust cos or mls qos trust dscp applies only for inbound packets. Ideas? Thanks! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Called Party display
Voice service voip No supplementary-service h225-notify CID-update Matthew Berry **Sent from my iPhone** Skype/Twitter: ciscovoiceguru Google Voice: +1 612 424 5044 On Jul 23, 2010, at 12:31, Brian Valentine bkvalent...@gmail.com wrote: I'm working on Vol2 Lab7 Task2.4. The task involves the following: HQ phone 2 dials 914158884343. Prefer to use TEHO to route the call out BR1. Local telco expects 7 digits. BR1 is an H323 gateway, so CUCM sends it 98884343. The gateway strips the 9 before sending to telco. Second choice gateway is the HQ gateway, which is MGCP. Local telco will expect 11 digits. CUCM would send the gateway 14158884343. Regardless of which gateway the call goes out the HQ Phone 2 display should say: To 4158884343. Got the call routing and redundancy down fine. That's works well enough. The problem is that no matter what I do, it seems to convert the display on HQ Phone 2 to match whatever digit manipulation was required by the egress gateway. The proctor guide says: The display on the Calling phone will be derived from the Route Pattern manipulation although the actual digits the UCM sends to the gateway is determined by the Route List/Route Group Called # transformations. So, I tried that. I tried doing all my digit manipulation on the RL details level and use the XX as the Called Party transformation on the Route Pattern level. Call goes through, but the HQ Phone 2 still displays To: 98884343. Next I tried setting the RL details to leave it as 415888 and used a Called Party Transformation Pattern at the gateway level to convert the call. I got the same result. Call succeeds. The display on HQ Phone 2 shows To: 98884343. What am I missing? Is this task possible? Thanks in advance, Brian ___ 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] Mobile Voice Access
Warren There should be a service parameter that toggles between using the gateway CSS or the device + remote destination profile CSS for call routing decisions. Yu could try playing with that. Matthew Berry **Sent from my iPhone** Skype/Twitter: ciscovoiceguru Google Voice: +1 612 424 5044 On Jul 19, 2010, at 20:12, Warren Heaviside (wheavisi) wheav...@cisco.com wrote: Experts, I’m configuring MVA and having the following issue: PSTN call comes into HQ H.323 G/W from 415-888-4343. Caller hears the VXML script prompt to enter the Remote Destination (4158884343) followed by pound, then a prompt for their PIN followed by #. The next prompt is to press 1# to place a call. After the number is dialed I can see the line appearance for HQ Phone x5002 go offhook but hear the Cisco reorder verbiage “your call cannot….”. The number unsuccessfully being dialed via MVA can be successfully dialed directly from the same HQ phone x5002. My MVA number is 5009 in CUCM. The POTS and VOIP dial peers are below. Thanks for looking. dial-peer voice 5009 pots service ccm incoming called-number 5009 no digit-strip dial-peer voice 5000 voip destination-pattern 5...$ voice-class codec 1 session target ipv4:10.10.210.11 dtmf-relay h245-alphanumeric no vad application service CCM http://10.10.210.10:8080/ccmivr/pages/IVRMainpage.vxml Warren Heavisidewheav...@cisco.com ENGINEER.CUSTOMER SUPPORT Phone: +1 408 853 7995 Office Hour 9 am - 5 pm Pacific Monday - Friday For corporate legal information go to: http://www.cisco.com/web/about/doing_business/legal/cri/index.html ___ 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] isdn plan
I set it for called and calling. Matthew Berry **Sent from my iPhone** Skype/Twitter: ciscovoiceguru Google Voice: +1 612 424 5044 On Jul 8, 2010, at 11:42 PM, Mark Holloway m...@markholloway.com wrote: Are you setting plan/type for both the called and calling numbers or just one of them? For example, if a task says the pstn provider wants the called party number type set and you set the plan/type for the called number, are you just leaving the calling portion set to CallManager or are you setting the plan/type for that as well? On Jul 7, 2010, at 11:43 AM, Berry, Matthew J. wrote: I make a habit of always setting the plan to ISDN. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.com -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Holloway Sent: Wednesday, July 07, 2010 1:40 PM To: OSL osl Subject: [OSL | CCIE_Voice] isdn plan When tasked with setting the call type to unknown, subscriber, national, or international, are you guys also setting the plan to isdn or are you just specifying the type and leaving the plan as unknown even though all the pstn access is isdn? ___ 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] ipphone does not looking for List.xml
Jeremy - Did you add the tftp-server commands to the CME router? *Matthew Berry* *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru On 7/6/2010 4:44 AM, jeremy co wrote: Hi, I'm trying to customize the background on my ipphone and downloaded List.xml and image and thumbnail to flash. Problem I faced from very begining is debug tftp does not show that ipphone looking for List XML at all. I reset the phone ,but same result. Jul 6 09:39:21.171: TFTP: Server request for port 49178, socket_id 0x66B007B4 for process 186 Jul 6 09:39:21.171: TFTP: read request from host 142.4.30.1(49178) via FastEthernet0/0.300 Jul 6 09:39:21.171: TFTP: Looking for CTLSEP0017E066C0CD.tlv Jul 6 09:39:21.171: TFTP: Sending error 1 No such file Jul 6 09:39:21.299: TFTP: Server request for port 49179, socket_id 0x66B007B4 for process 186 Jul 6 09:39:21.299: TFTP: read request from host 142.4.30.1(49179) via FastEthernet0/0.300 Jul 6 09:39:21.299: TFTP: Looking for SEP0017E066C0CD.cnf.xml Jul 6 09:39:21.307: TFTP: Opened flash:/its/vrf1/XMLDefault7961.cnf.xml, fd 0, size 1099 for process 186 C2801(config-ephone)# Jul 6 09:39:21.307: TFTP: Sending block 1 (retry 0), socket_id 0x66B007B4 Jul 6 09:39:21.307: TFTP: Received ACK for block 1, socket_id 0x66B007B4 Jul 6 09:39:21.307: TFTP: Sending block 2 (retry 0), socket_id 0x66B007B4 Jul 6 09:39:21.311: TFTP: Received ACK for block 2, socket_id 0x66B007B4 Jul 6 09:39:21.311: TFTP: Sending block 3 (retry 0), socket_id 0x66B007B4 Jul 6 09:39:21.311: TFTP: Received ACK for block 3, socket_id 0x66B007B4 Jul 6 09:39:21.311: TFTP: Finished flash:/its/vrf1/XMLDefault7961.cnf.xml, time 00:00:00 for process 186 C2801(config-ephone)# Jul 6 09:39:22.723: TFTP: Server request for port 49180, socket_id 0x66B007B4 for process 186 Jul 6 09:39:22.723: TFTP: read request from host 142.4.30.1(49180) via FastEthernet0/0.300 Jul 6 09:39:22.723: TFTP: Looking for English_United_States/mk-sccp.jar Jul 6 09:39:22.723: TFTP: Sending error 1 No such file Jul 6 09:39:22.867: TFTP: Server request for port 49181, socket_id 0x66B007B4 for process 186 Jul 6 09:39:22.867: TFTP: read request from host 142.4.30.1(49181) via FastEthernet0/0.300 Jul 6 09:39:22.867: TFTP: Looking for United_States/g3-tones.xml Jul 6 09:39:22.867: TFTP: Sending error 1 No such file C2801(config-ephone)# Jul 6 09:39:23.407: %IPPHONE-6-REG_ALARM: 22: Name=SEP0017E066C0CD Load= SCCP41.8-3-3S Last=Reset-Reset Jul 6 09:39:23.439: %IPPHONE-6-REGISTER: ephone-5:SEP0017E066C0CD IP:142.4.30.1 Socket:3 DeviceType:Phone has registered. Cheers, Jeremy ___ 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] Layer 2 Overhead Size on Frame Relay
Graham - IPexpert has several quality explanations for the size in bytes of each of these layer two methods. However, when approaching the QoS section of the CCIE Voice lab, I personally have opted to only use the QoS SRND. During last years, Ask the Expert forum, Ben Ng stated that he always uses the QoS SRND. Since he writes the labs for the voice track, I've decided to use the same standard he does. However, I believe it's been said that the proctors allow for a certain variance between the stated solution and what the candidate enters in his configuration. *Matthew Berry* *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru On 7/6/2010 3:25 AM, Graham Hopkins wrote: This question keeps cropping up so I thought I'd share my findings on this: QoS SRNDham - MLP - 13 bytes FR - 4 bytes FRF.12 - 8 bytes UC SRND MLP - 10 bytes FR - 4 bytes FRF.12 - can't find it. Looking at the standards RFC 1990 for MLP Long Sequence Number Format 10 bytes Short Sequence Number Format 8 bytes So together with the UC SRND I assume Cisco use the Long Sequence Number Format and would use the 10 bytes figure FRF.12 Seems to have options for example on Cisco the End-to-End Fragmentation and Switched PVC Fragmentation formats are different: However in Cisco Press - Cisco Frame Relay Solutions Guide - I find (Figure 16.3) 2 bytes FR Header 2 bytes UI and NLPID ( Network Layer Protocol Identifier) 2 bytes Fragmentation Header 2 bytes FCS Total 8 bytes which matches the QoS SRND Normal FR is a 2 byte header and 2 byte FCS giving 4 bytes as in both SRNDs So summary - MPLP = 10 bytes, FRF.12= 8 bytes FR = 4 bytes Any other options welcome. Regards Graham Hopkins ___ 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] Multicast MoH Verification
I have setup multicast music-on-hold at my BR-2 CME site. When I call the PSTN phone from BR2 and place it on hold, the PSTN phone hears Cisco's lovely MOH. When I call from BR2-Phone1 to BR2-Phone2, I hear silence. I am trying to determine if this is because I am running on Proctor Labs. Another notable anomaly is that show ccm-manager music-on-hold displays 0 multicast sessions. Ideas? I am labbing for another 5-6 hours. Hope to hear back from someone soon! Thanks! Matthew ccm-manager music-on-hold ... telephony-service max-ephones 40 max-dn 40 no-reg primary ip source-address 10.10.202.1 port 2000 moh music-on-hold.au multicast moh 239.1.1.1 port 16384 route 10.10.202.1 10.10.110.3 (additional commands omitted) BR2-RTR# debug ephone moh Jul 7 03:16:13.073: MoH route If Vlan400 ETHERNET 10.10.202.1 via ARP Jul 7 03:16:13.073: MoH route If Loopback0 46 10.10.110.3 via 10.10.110.3 BR2-RTR# Jul 7 03:16:18.177: MoH route If Vlan400 ETHERNET 10.10.202.1 via ARP Jul 7 03:16:18.177: MoH route If Loopback0 46 10.10.110.3 via 10.10.110.3 BR2-RTR# Jul 7 03:16:23.317: MoH route If Vlan400 ETHERNET 10.10.202.1 via ARP Jul 7 03:16:23.317: MoH route If Loopback0 46 10.10.110.3 via 10.10.110.3 BR2-RTR#show ccm-manager music-on-hold Current active multicast sessions : 0 -- *Matthew Berry* *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUE and CUCM intergration issue
Try restarting Call Manager and CTI Manager services on the Sub and Pub. *Matthew Berry* *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru On 7/5/2010 4:47 AM, Miron Kobelski wrote: Hi Kevin, have you associated the RP with VM profile? regards kobel On Mon, Jul 5, 2010 at 11:11 AM, Hobson Kevin kevin.hobson2...@ntlworld.com mailto:kevin.hobson2...@ntlworld.com wrote: Hi all, I am having real issues with this. The CUE just refuses to register to the UCM. Below is what i have done so far to try and get this working: ___ 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] Vol2 Lab 8 2.3 and 2.8
Bruce - Have you ran a debug gatekeeper main 10? Could you include this debug in this email thread? My guess is that you have a 1# technology prefix coming into CUCM with the 5002 (1#5002) and your H.225 trunk is set to ALL for significant digits. For such a case, you would need to set your incoming digits to 4 or add a translation pattern that would match 1#. with a DDI-PreDot. The fast busy would make sense if your 3002 number is a SIP phone since the CUCM Annunciator is not supported for SIP. If you are calling from SCCP phones at Site C, then you'd be dealing with another issue. Since your voice ccapi inout debugs return an unallocated number, this leads me to believe that your underlying infrastructure is setup correctly; your call routing, however, is not. Let me know what you find out. *Matthew Berry* *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru On 7/2/2010 6:17 PM, Bruce Clapp wrote: Hello, I am having trouble with calls coming into UCM when using SIP early offer. I have inbound fast start selected on my trunk. Outbound calls from HQ and BR1 to BR2 work fine. When I call 5002 from 3002, I get a fast busy. If I look at the voip ccapi inout debugs on HQ-RTR when I try calling 5002 from 3002, It seems to indicate the I am reaching an unallocated number. I checked the css on the trounk for inbound calls, and it has visibility of the phones and route patterns for the pstn. I have the MTP selected, and a transcoder available to the trunk. Thoughts? *Bruce Clapp* Senior Systems Engineer Right! Systems Inc. 2600 Willamette DR NE Lacey WA 98516 +1.360.528.4070 Single Reach Number +1.360.956.0336 Fax bcl...@rightsys.com BLOCKED::mailto:bcl...@rightsys.com E-MAIL PRIVILEGED INFORMATION This email message is for the sole use of the intended recipient's and may contain confidential and privileged information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. If you are the intended recipient, please be advised that the content of this message is subject to access, review and disclosure by the sender's Email System Administrator. ___ 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
Sean - Did you figure this one out? *Matthew Berry* *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru On 7/2/2010 11:12 AM, sean hurricane wrote: Ashar, I have tried your configuration and it does not satisfy the requirementi will try Kobel's configuration later.. On Fri, Jul 2, 2010 at 11:33 AM, Ashar Siddiqui siddas...@gmail.com mailto:siddas...@gmail.com wrote: It will not. You are not configuring ephones, you are just configuring privacy thing. Ash *From:* kobel [mailto:findko...@gmail.com mailto:findko...@gmail.com] *Sent:* 02 July 2010 16:00 *To:* Ashar Siddiqui *Cc:* sean hurricane; ccie_voice@onlinestudylist.com mailto:ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] SRST hi, doesn't this break the 2nd requirement? I've never tried this, but I would configure ephone-template and assign it to srst via srst ephone template command of telephony-services. sccp group should indeed be configured with srst as the 3rd option. regards kobel On Fri, Jul 2, 2010 at 4:29 PM, Ashar Siddiqui siddas...@gmail.com mailto:siddas...@gmail.com wrote: Sean, Do srst auto-prov none and then just create ephones (as many as required) and put the following in there: Ephone 1 No privacy ! Ephone 2 No privacy ! You will need to do all the basic requirements for Cbarge like conference hardware, sdspfarm units etc and configuring dspfarm with telephony-service address on third priority if the requirement is that Cbarge is working during normal mode as well. Give it a go and let us know how it works. Ash ___ 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] UCCX Script question
Phil - The script looks good to me, but I haven't verified it in a lab. I couldn't determine from your email if there was an actual problem with the script or not. When you loaded it and ran tests, did you observe any unexpected behavior? Thanks! *Matthew Berry* *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru On 7/2/2010 12:20 PM, Phillip Day wrote: Hi, I have created a fairly complex script for a customer, and the only part I can't test yet is the queueing. I have attached a screenshot of what happens when a call is queued. Can anyone see any reason why this wouldn't work. The script has been validated, and we are about to add some resources to the CSQs, I'm just wondering if I could have missed anything here? The idea is that roughly every 15 second the user gets a prompt to say a customised hold message, then they get another to say where they are in the queue, and then another to say the estimated wait time. If they are 10th or more in the queue, there is no prompt for the wait time but the other two should play. Anyone see any problems with this?? Thanks in advance Phill __ This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. This footnote also confirms that this email message has been swept by a content checking tool for the presence of computer viruses. Nettitude Limited is a Company registered in England Registered Address Nettitude Limited, Fosse House, Fosse Way, Leamington Spa, Warwickshire, CV31 1XG Company Registration Number: 4705154 VAT Number: 812 4539 44 www.nettitude.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] Music on hold from router flash (Piano music)
Remember to include a loopack address if you want MMOH to play for the PSTN callers. On 7/2/10, Randall Saborio ill2...@gmail.com wrote: For HQ, streaming from CM, it would make sense to increase the Max Hops. The way I understand it, the max hops would account for any extra interface that the stream has to go through. So for example, from CM towards default gateway on servers VLAN is 1 hop, then towards voice vlan default gateway will be 2 hops. For streaming from the HQ and BR1 router, I am suggest checking the show run and make sure that you include the interfaces through which the moh will be streamed, like this: call-manager-fallback multicast moh 239.1.1.1 port 16384 route 10.10.201.1 Where the route address is the voice vlan interface on the router. You can include many addresses for every interface you want to route the multicast through. HTH. On Fri, Jul 2, 2010 at 1:43 PM, Mark Holloway m...@markholloway.com wrote: Is your Site B router MGCP or H323? With an H323 gateway I could get the router to stream the local piano music while the MoH server is set to one hop in UCM. With an MGCP gateway I couldn't get this to work and it always streams from UCM unless the router is in SRST mode then it plays piano music. I am also using a home lab. I tried to isolate why this wasn't working but could never come up with a root cause. On Jul 2, 2010, at 7:52 AM, Afzal Bhutta wrote: Sorry Folks not providing details in first attempt. Thanks for all and special thanks to Matthew Berry and Randall Saborio for their interest and figured out this issue. Let’s make thing more understandable. I am working in my home lab. I am trying to spoof call manager. My target is to get music from router flash for HQ and for siteB not from call manager. Call manager is configured as I explain below. I have adjusted my ServerMax Hops to 15 for the M.cast it is working fine for HQ and SiteB but I am hearing music from Call manger not from router flash (Piano music) What I performed on the routers. I have enabled Muticast-routing on HQ and site B I have use IP pim dense mode commands on Voice-vlan interfaces, Loopback0, and serial interfaces which are connected to frame relay (WAN links) both for HQ and Site B. CCM-manager music on hold command is also on both sites. Site B is providing SRST. SRST is configured using telephony command. Troubleshooting: When I adjusted ServerMax Hops to 2 still I can hear music from call manager. I tested it in this way. Call from HQ to SiteB, HQ-ph is put on hold and I can hear music from the router flash (Piano music) If site B is put on hold I can hear call manger music (Actually it should be from router flash- Am I right?) When I adjusted my ServerMax Hops to 1 for the M.cast it is not working I can not hear any music just silence even no beeps. Even within HQ phone, when they call each other I put one of them on hold I can not hear any music not from call manger nor from router flash. Yes I can hear music from router flash when I call from HQ to Site B with adjusted my ServerMax Hops to 2 and put HQ phone on hold but when I put hold for SIteB Phone nothing I can hear completely silent even no beeps Here is Call manager config details, MOH is multicast on 239.1.1.1 port 16384.Allow multicasting is enable on CUCM-PUB. CallManager MoH Server Increment Multicast on = IP Address CallManager MoH ServerMax Hops = 1 MOH Audio Source:? SampleAudioSource (1) = Allow Multicasting In Media Resource Group =? Use Multicast for MOH Audio (This is enable) CME is completely separate side,It is not participating in this Scenario. IP Voice Media Streaming App is enabled for G729 and G722 in service parameter.(Cisco IP Voice Media Streaming App = 711 uulaw and 729 Annex A selected) I have MOH region with G711ulaw enable with all other region with codec G711ulaw. HQ device pool using MRGL SiteB device pool using MRGL MRGL contains MOH-PUB-MULTI-RG All phones within site (Intra-site) using G711ulaw where as between site (Inter-site) they are using G729ulaw. ___ 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 -- Sent from my mobile device ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MGCP
Try a no mgcp / mgcp on the gateway. When an MGCP router registers with CUCM for the first time, it will use the serial interface. If it doesn't work, please send your config. *Matthew Berry* *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru On 7/2/2010 8:36 AM, Duncan Hamilton-Walker wrote: Hi All, Why would my BR1-RTR being registering with UCM on the serial sub interface 10.10.111.2 When i have configured it to use the Loopback0 for source packets.. 10.10.110.2 Plus i think is not helping with making any call out of the gateway !! Thanks Duncan ___ 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] Lab 5 Volume 2 LLQ sizing and RSVP CAC
Daniel, RSVP always calculated bandwidth based on the worst-case scenario for the first call and normal scenario for all the remaining calls. The easiest way to calculate this is to bump up your RSVP bandwidth to something really high like 1000. Then place a call through the RSVP call agents with debug ip rsvp signalling turned on. You will see what RSVP is using for worst case and normal call scenarios. Then just recalculate based on those values. It's must easier than remembering. *Matthew Berry* *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru On 6/27/2010 3:16 PM, Daniel Berlinski wrote: Thanks Kobel for your explanation. It does make sense to me that small voice packets do not grow in size to the point of being fragmented. The only thing that I'm not too sure is whether or not this would be something that Cisco would be expecting to see as a valid answer if such a question was asked in the exam. I guess I would ask the proctor because the lab exam is usually far from reality. Hello Mouhammad I disagree with your statement Finally, I know that both LLQ and ip rsvp bandwith values must be identical and calculated as = (N-1) calls at 20 mSec + 1 call at 10 mSec Why would you always match those two values? Are you calculating these with or without layer 2 overhead? There is an example in the UCM 7 SRND page 3-64 which describes RSVP calculation examples without taking any layer 2 overhad into account. There is a note on page 3-64 that states Unified CM does not include SRTP overhead or the L2 overhead int he RSVP reservation. and then it says that the layer 3 IP rsvp bw statement must take into account any SRTP traffic and the L2 priority queue must also be over-provisioned if SRTP is present. How do you guys interpret this and what should we do to get those precious points in the exam?? 2010/6/28 Mouhammad Nasser engnasse...@hotmail.com mailto:engnasse...@hotmail.com Hi Kobel, The worst case takes a place upon the initialization of each RSVP call calculation, CUCM 7.0 LLD refers that amond N calls, it is recommended to calculate call number N as worst case, so it always succeeds (written in P.3-64 Configuration Recommendation) Regarding the number of bytes in FRF.12 header, do you recommend we always consider it a 4 Bytes? It is not mentioned in CUCM LLD, and I saw it fixed at 8 bytes in QoS LLD, I think it is better to go with 8 I don't know. I hope someone from IPExpert to explain this more, Amy: we shall be waiting for your kind reply here Finally, I know that both LLQ and ip rsvp bandwith values must be identical and calculated as = (N-1) calls at 20 mSec + 1 call at 10 mSec Thank you a lot in advance Hotmail: Trusted email with powerful SPAM protection. Sign up now. https://signup.live.com/signup.aspx?id=60969 ___ 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] Music on Hold
That's a really good suggest, Daniel, and something that I completely forgot about. Afzal, remember that if you're using ProctorLabs you'll need to bump that up quite a bit due to the EZVPN connection. Try something like 10. That should cover your needs. *Matthew Berry* *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru On 6/28/2010 5:11 PM, Daniel Berlinski wrote: Hello Afzal From what you told us it appears that you need to adjust your ServerMax Hops to a value greater than 1 in order for the Mcast stream to reach your branch phones. have you tried doing that? On Tue, Jun 29, 2010 at 7:05 AM, Afzal Bhutta azhar.bhu...@gmail.com mailto:azhar.bhu...@gmail.com wrote: Hello, Here is some more details, MOH is multicast on 239.1.1.1 port 16384.Allow multicasting is enable on CUCM-PUB. CallManager MoH Server Increment Multicast on = IP Address CallManager MoH ServerMax Hops = 1 MOH Audio Source: SampleAudioSource (1) = Allow Multicasting In Media Resource Group = Use Multicast for MOH Audio (This is enable) CME is completely separate side,It is not participating in this Scenario. IP Voice Media Streaming App is enabled for G729 and G722 in service parameter.(Cisco IP Voice Media Streaming App = 711 uulaw and 729 Annex A selected) I have MOH region with G711ulaw enable with all other region with codec G711ulaw. HQ device pool using MRGL SiteB device pool using MRGL MRGL contains MOH-PUB-MULTI-RG All phones within site (Intra-site) using G711ulaw where as between site (Inter-site) they are using G729ulaw. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://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] GK CUBE behaviour
Nauofal - Make sure your gatekeeper trunk in UCM has these settings: - Media Termination Point Required (checked) - Wait for Far End H.245 Terminal Capability Set (unchecked) - MRGL set with an MRG that has a SW MTP setup on the HQ router - No outbound FastStart - Inbound SlowStart only required for SIP phones (due to Early Offer) Make sure you setup a software MTP on the HQ gateway: sccp local FastEthernet0/0.20 sccp ccm 10.10.210.10 identifier 2 priority 2 version 5.0.1 sccp ccm 10.10.210.11 identifier 1 priority 1 version 5.0.1 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate ccm 2 priority 2 associate profile 20 register HQ-SW-MTP ! dspfarm profile 20 mtp codec pass-through codec g729r8 maximum sessions software 5 associate application SCCP *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru On 6/24/2010 6:19 AM, naoufal.kerboute wrote: Hi all, I'm working on lab2 Vol10 section gatekeeper, and I have a problem when calling from UCM to BR2 via the CUBE. The call routed correctly but when I answer from BR2, UCM phones still ringing (SIP and SCCP). I remember I had the same problem and it was related to sip phones, when I had to disable the TCS wait on the trunk. Below my HQ-RTR config: voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip ! ! interface Loopback0 ip address 10.10.110.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id VIA ipaddr 10.10.110.1 1719 h323-gateway voip h323-id HQ-RTR ! sccp local FastEthernet0/0.20 sccp ccm 10.10.200.3 identifier 1 version 5.0.1 sccp ! sccp ccm group 1 bind interface FastEthernet0/0.20 associate ccm 1 priority 1 associate profile 2 register cube-xcoder ! dspfarm profile 2 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 6 associate application SCCP ! ! dial-peer voice 1 pots incoming called-number . direct-inward-dial port 0/0/0:23 ! dial-peer voice 2 voip incoming called-number 3 ! dial-peer voice 3000 voip destination-pattern 3... session target ras codec g711ulaw ! ! gateway ! ! ! gatekeeper zone local UCM ipexpert.com zone local UCME ipexpert.com outvia VIA zone local VIA ipexpert.com zone prefix UCM 1... zone prefix UCME 3... zone prefix UCM 5... gw-type-prefix 1#* default-technology no shutdown ! ! telephony-service sdspfarm units 1 sdspfarm transcode sessions 8 sdspfarm tag 1 cube-xcoder max-ephones 1 max-dn 1 ip source-address 10.10.200.3 port 2000 max-conferences 8 gain -6 moh music-on-hold.au multicast moh 239.1.1.1 port 16384 route 10.10.110.1 10.10.200.3 transfer-system full-consult ! Any Idea? note: from BR2 to UCM calls succeeds and answered on both side. Regards Naoufal ___ 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] Streaming MoH from Router Flash
Can you post your config? I had a similar issue earlier that was caused by omitting the ip source command under telephony-service or call-manager-fallback. By doesn't work, do you mean that you hear dead air or tone on hold? *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru On 6/23/2010 10:04 AM, Mark wrote: If PUB is configured for multicast MoH, 239.1.1.1, port 16384, increment by IP, and I need to stream MoH from router flash on both BR1 and BR2, I'm having difficulty getting it to work on BR2. I have an MRG called PUB_MCAST_MoH and I've assigned it to both MRGL's MRGL_BR1 and MRGL_BR2 which are assigned to their respective Device Pools. On the BR1 router, under call-manager-fallback, I have set 'moh multicast 239.1.1.1 port 16384 routevoice vlan ip loopback ip' and it's working. If I repeat the same process on BR2 it doesn't work. I've verified ip multicasting is set and dense mode is set. Any suggestions? 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
[OSL | CCIE_Voice] Gateway Registration to Gatekeeper
Here's my question, plain and simple: *What elements are required for a gateway to register with a gatekeeper?* I have been reading through various Cisco documents and the CUCM SRND GK Address Resolution on ARQ and have not been able to find a clear-cut list of requirements for registration. I'm aware that it's normal to enter a tech-prefix on the gateway and zone prefixes on the gatekeeper. But in this question I'm wanting to go beyond the typical and understand what is possible. So far, here are the ways I've seen that will return an ACF: 1. Tech Prefix match 2. Static address registration 3. Automatic address registration of ephone-dns Any ideas on this? -- *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Gateway Registration to Gatekeeper [Clarification]
All - Sorry, guys. I'm mixing up two different things. I'm not talking about gatekeeper registration. I'm talking about call routing through the gatekeeper and what information is required for the call go through. I can setup tech-prefixes and zone locals on the gateway and gatekeeper respectively. What I am looking for is additional ways to route these calls if I am restricted somehow from using the standard setup. *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru On 6/21/2010 7:51 AM, Matthew Berry wrote: Here's my question, plain and simple: *What elements are required for a gateway to register with a gatekeeper?* I have been reading through various Cisco documents and the CUCM SRND GK Address Resolution on ARQ and have not been able to find a clear-cut list of requirements for registration. I'm aware that it's normal to enter a tech-prefix on the gateway and zone prefixes on the gatekeeper. But in this question I'm wanting to go beyond the typical and understand what is possible. So far, here are the ways I've seen that will return an ACF: 1. Tech Prefix match 2. Static address registration 3. Automatic address registration of ephone-dns Any ideas on this? -- *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Lab 8 - RSVP Remarking to CS1
In Question 5.6, we are asked to police traffic that was not admitted via RSVP-enabled Locations-CAC. The Proctor Guide provides a solution that identifies such traffic as CS1: class-map match-all RSVP-REMARK match ip dscp cs1 ! policy-map RSVP-REMARK-POLICY class RSVP-REMARK police rate percent 33 As a follow-up to last night's lab experience, I am trying to determine how the traffic is marked to CS1. Initially, I thought that this would be done in IOS, but after five hours of sleep I'm thinking that this is set in CUCM with the following CCM Service Parameter: DSCP for Audio Calls when RSVP Fails = CS1(precedence 1) DSCP 001000 The default is default DSCP 00 This step is not listed in the Proctor Guide, but I believe this will correctly remark failed RSVP traffic as it leaves CUCM enroute to BR1 phones over the RSVP agent configured on the gateways. Is this correct? -- *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 8 - RSVP Remarking to CS1
This solution was in the Proctor Guide after all. I think I might have been using an older version of the guide that did not have this solution in it. Thanks. *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 6/16/2010 5:54 AM, Matthew Berry wrote: In Question 5.6, we are asked to police traffic that was not admitted via RSVP-enabled Locations-CAC. The Proctor Guide provides a solution that identifies such traffic as CS1: class-map match-all RSVP-REMARK match ip dscp cs1 ! policy-map RSVP-REMARK-POLICY class RSVP-REMARK police rate percent 33 As a follow-up to last night's lab experience, I am trying to determine how the traffic is marked to CS1. Initially, I thought that this would be done in IOS, but after five hours of sleep I'm thinking that this is set in CUCM with the following CCM Service Parameter: DSCP for Audio Calls when RSVP Fails = CS1(precedence 1) DSCP 001000 The default is default DSCP 00 This step is not listed in the Proctor Guide, but I believe this will correctly remark failed RSVP traffic as it leaves CUCM enroute to BR1 phones over the RSVP agent configured on the gateways. Is this correct? -- *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Real time tracing of IP phone activity
Paul - Try fail tail activelog /cm/trace/ccm/sdi recent. If you look at your command below, you are using /cm/trace/cmi/sdi. *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 6/16/2010 6:18 AM, Paul Dardinski wrote: All, Is there something beyond enabling trace level in Servicability that is required to get this to work? I have enabled serv level to significant and I can see the SDI traces appear in RTMT (with delay), but the “file tail activelog /cm/trace/cmi/sdi recent” isn’t working for me. I get a short output, but beyond that it never feeds anything else. I have forced the pub to be primary just to confirm that it would be the call agent for the output. Paul (#16842 RS/Sec) admin:file tail activelog /cm/trace/cmi/sdi recent 06/07/2010 00:30:26.957 CMI|DB: Str[ParamValue]=[1]|CLID::StandAloneClusterNID::10.1.200.20LVL::AllMASK:: 06/07/2010 00:30:26.957 CMI|DB: Int[tkParam]=[3]|CLID::StandAloneClusterNID::10.1.200.20LVL::AllMASK:: 06/07/2010 00:30:26.957 CMI|DB: MoveNext() EOF: TRUE|CLID::StandAloneClusterNID::10.1.200.20LVL::AllMASK:: 06/07/2010 00:30:26.957 CMI|DB: IsEOF(): TRUE|CLID::StandAloneClusterNID::10.1.200.20LVL::AllMASK:: 06/07/2010 00:30:26.957 CMI|DB: ~CFastAccess(ProcessConfig_EnterpriseWide)|CLID::StandAloneClusterNID::10.1.200.20LVL::AllMASK:: 06/07/2010 00:30:26.961 CMI|DB: CFastAccess(CallManager)|CLID::StandAloneClusterNID::10.1.200.20LVL::AllMASK:: 06/07/2010 00:30:26.961 CMI|SQL[select cm.pkid, cm.name, cm.fkprocessnode, pn.name as processnodename, cm.ctiid from callmanager as cm, processnode as pn where cm.fkprocessnode = pn.pkid and cm.fkprocessnode = '551fe708-709c-45ad-9b93-eb4687bd2ca6']|CLID::StandAloneClusterNID::10.1.200.20LVL::AllMASK:: 06/07/2010 00:30:26.977 CMI|DB: MoveNext() EOF: FALSE|CLID::StandAloneClusterNID::10.1.200.20LVL::AllMASK:: 06/07/2010 00:30:26.977 CMI|DB: IsEOF(): FALSE|CLID::StandAloneClusterNID::10.1.200.20LVL::AllMASK:: 06/07/2010 00:30:26.977 CMI|DB: Guid[fkprocessnode]=[551fe708-709c-45ad-9b93-eb4687bd2ca6]|CLID::StandAloneClusterNID::10.1.200.20LVL::AllMASK:: *From:* ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *kobel *Sent:* Tuesday, June 15, 2010 7:18 AM *To:* Matthew Berry *Cc:* ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] Real time tracing of IP phone activity Just a side note - I also use those two commands, but with a little modification: file tail activelog /cm/trace/cmi/sdi recent this makes the CLI to choose the most recent file (no need to type in the filename yourself). RTMT is such a waste of time, when it comes to traces ;) BTW, the most useful are SDI traces (SDLs are less readable and are used for inter-ccm communications - I never use them in lab). It's easy to remember - SD-III like IIIncredibly useful traces :D regards kobel On Tue, Jun 15, 2010 at 1:00 PM, Matthew Berry ciscovoiceg...@gmail.com mailto:ciscovoiceg...@gmail.com wrote: I would turn on detailed tracing through CUCM Serviceability and then monitoring the SDL or SDI traces (I always forget which one) through the CUCM CLI. It's the best way I know how. file tail activelog /cm/trace/cmi/sdl file tail activelog /cm/trace/cmi/sdi ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Lab 8 - Question 5.5 // RTP Priority Queue
In question 5.5 we are asked to create a priority queue of 128 kbps for RTP traffic between HQ and BR2. The Proctor Guide told me to set: interface Virtual-Template 200 no service-policy output AutoQoS-Policy-UnTrust ip rtp priority 16384 16383 128 I would have normally set this with a class-map and policy-map. Are we setting this priority queue under the Virtual-Template because of the no-MQC restriction in the question? Now that I am writing this, I am pretty sure this is the reason, but I want to verify. -- *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] QOS FRF.12 MLPP
Kobel, In my opinion, you should only retain the frame-relay ip rtp header-compression under the frame-relay DLCI if you are asked to compress the rtp packets. Because we're dealing with a slow-speed link, auto qos tries to be helpful by adding in this command. My general stance when it comes to answering the QoS lab questions is to only configure what they ask you to setup. Using auto qos is helpful to rough-in a configuration, but leaving in unnecessary elements does not demonstrate a mastery of the knowledge you are being tested on. I will provide another example: When you type auto qos voip several classes will be created. One of those classes, called something like remark, will set DSCP values on so-called rogue traffic masquerading as media or signaling traffic. If the question does not ask you to perform that task, you'll want to remove the remark class. I'm not sure if this helps, but it's my take on the subject. My guess is that the lab would be specific whether they wanted class-based cRTP or not. *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 6/15/2010 4:23 AM, kobel wrote: Also, after the Auto QOS generates a lot of classes etc. We do edit few things here and there. Just wanted to confirm that is it a good practice to remove rtp header compression? I use to remove it always but now I am getting conflicting feedback that should we remove it or not? interface Serial0/2/0.1 point-to-point bandwidth 256 frame-relay interface-dlci 301 CISCO class AutoQoS-FR-Se0/2/0-301 auto qos voip trust *frame-relay ip rtp header-compression* I would appreciate any input in this regard. you can configure cRTP in two ways. if the task doesn't explicitly ask for CB cRTP, I keep auto qos config - why waste time? I'm not aware of any drawback of this method. ___ 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] Real time tracing of IP phone activity
I would turn on detailed tracing through CUCM Serviceability and then monitoring the SDL or SDI traces (I always forget which one) through the CUCM CLI. It's the best way I know how. file tail activelog /cm/trace/cmi/sdl file tail activelog /cm/trace/cmi/sdi *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 6/15/2010 4:23 AM, ShinGei Yong wrote: Hi, I'm trying to do a real time tracing of ip phone activity, for example; when the phone goes off hock, the line seized, CUCM sending the signaling and tone etc... I'm using RTMT -- Real time tracing -- View real time data to do so but unsuccessful. Anyone know which services should be select when using RTMT for such action? Thanks shingei. ___ 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] CBarge in SRST ~(Again)
Angel - I think you are right. The only way I can see of configuring privacy on/off would be through the ephone section itself. Privacy isn't an option with an ephone-template, otherwise you could have set it there. You could possibly set no privacy under telephony-service, but that would be a global setting. I am not at my lab right now so I cannot verify if that would actually propagate down to SRST-provisioned phones. *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 6/15/2010 3:37 AM, Angel Perez wrote: Hi: I was wondering how can you add privacy/privacy off to the ephone if you are setting srst auto none? The only way I can imagine is changing from srst auto all to auto none once the ephone are configured. Correct me if i'm wrong thanks Date: Mon, 14 Jun 2010 18:06:15 +0100 Subject: Re: [OSL | CCIE_Voice] CBarge in SRST ~(Again) From: cci...@gmail.com To: gorr...@hotmail.com CC: ccie_voice@onlinestudylist.com Hello Angel, Yes I made it work..its been quite few days now.. I just explicitly included privacy off commands under ephones and it worked. There is no need for srst auto prov all and dialpeer hunt 3 etc... hth On Mon, Jun 14, 2010 at 3:20 PM, Angel Perez gorr...@hotmail.com mailto:gorr...@hotmail.com wrote: Hi: Did you manage to make this work? Finally I got some time to relab it, if you are interested let me know and I'll post my working config thx Hotmail: Free, trusted and rich email service. Get it now. https://signup.live.com/signup.aspx?id=60969 Hotmail: Trusted email with powerful SPAM protection. Sign up now. https://signup.live.com/signup.aspx?id=60969 ___ 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] Configuring H.323 Call Preserve
When configuring call preservation for an H.323 gateway, I am using the following command: *voice service voip h323 call-preserve* As soon as I hit ENTER, the IOS spits back this warning/notice to me: *Warning: Configuring media inactivity detection to avoid hung calls is highly recommended.* Does anyone know what I need to do in order to configure media inactivity detection? I want to make sure that I am entering the proper commands to ensure that H.323 call preservation is enabled. -- *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] OSPF Error CUE Module
I am getting an odd OSPF error after having configured my service-engine for the CUE module: *Jun 14 05:46:22.401: %OSPF-4-NO_IPADDRESS_ON_INT: No IP address for interface Service-Engine0/0* Everything appeared to function properly even with this error being reported. Below is my example config that I use to configure the CUE module's IP and connectivity: *interface FastEthernet 0/0.101 ip address X.X.X.X 255.255.255.0 interface Service-Engine 0/0 ip unnumered FastEthernet 0/0.101 service-module ip address X.X.X.X 255.255.255.0 service-module ip default-gateway Y.Y.Y.Y no shut ip route X.X.X.X 255.255.255.255 Service-Engine 0/0 * -- *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Configuring H.323 Call Preserve
Thanks, Angel! *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 6/15/2010 6:50 AM, Angel Perez wrote: /Allow Peer to Preserve H.323 Call/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] OSPF Error CUE Module
I figured as much, but it's always better to run it by your fellow egg heads before assuming. Thanks, brutha' *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 6/15/2010 6:52 AM, Angel Perez wrote: Hi: This is becouse you are setting ip unnumbered, there is another method with ip address, with it you won't get this error But the error it's just cosmetic hth Date: Tue, 15 Jun 2010 06:48:58 -0500 From: ciscovoiceg...@gmail.com To: ccie_voice@onlinestudylist.com; ciscovoiceg...@gmail.com Subject: [OSL | CCIE_Voice] OSPF Error CUE Module I am getting an odd OSPF error after having configured my service-engine for the CUE module: *Jun 14 05:46:22.401: %OSPF-4-NO_IPADDRESS_ON_INT: No IP address for interface Service-Engine0/0* Everything appeared to function properly even with this error being reported. Below is my example config that I use to configure the CUE module's IP and connectivity: *interface FastEthernet 0/0.101 ip address X.X.X.X 255.255.255.0 interface Service-Engine 0/0 ip unnumered FastEthernet 0/0.101 service-module ip address X.X.X.X 255.255.255.0 service-module ip default-gateway Y.Y.Y.Y no shut ip route X.X.X.X 255.255.255.255 Service-Engine 0/0 * -- *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com mailto:ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 Hotmail: Powerful Free email with security by Microsoft. Get it now. https://signup.live.com/signup.aspx?id=60969 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME Calling Name
I think it's clip strip under the dial peer. At the movies so I can't verify on my lab unless there's a killer IPX vRack app for my iPhone. ;) Matthew Berry **Sent from my iPhone** Skype/Twitter: ciscovoiceguru Google Voice: +1 612 424 5044 On Jun 15, 2010, at 11:56 AM, ccievoice daniyal.vo...@gmail.com wrote: Hi could some one pls help to resolve this issue in CME i don't want send the Calling Name on specific dial-peer but Number suppose to go under D channel i have configured Isdn out display ie that affecting on all calls but requirement is that i just want to block or restrict one person/ dial-peer to don't show the calling Name comments/advise appreciated Dani ___ 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] Lab 8 - iLBC Locactions Setting
In Lab 8, you are asked to configure iLBC between HQ and BR1 with RSVP CAC on top of that. The Proctor Guide tells me to set the Link Loss Type under CUCM SYSTEM LOCATIONS to Lossy. However, all of my testing to date seems to demonstrate that the lossy setting does not affect whether iLBC is used between endpoints. I am wondering what the reason is for setting this option and whether it is necessary to complete the requirements of the question. -- *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Lab 8 - MeetMe Hardware Conferencing Resource
In the Proctor Guide for Lab 8, I am directed to setup a hardware transcoder for the purpose of facilitating a 10-party MeetMe conference. However, I seemed to be able to get everything setup properly without a hardware conferencing resource. In fact, I have done many customer implementations with MeetMe, none of which required hardware conferencing resources. The only reason I can think of is the requirement that HQ, BR1, and external PSTN callers be allowed to join the call. Does such a request necessitate the use of hardware conferencing? -- *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] VMware Connectivity and VLAN Interface Issues
Has anyone else experienced issues with Pod 12 on PL? I am using that pod right now and have two issues that prevent me from moving forward. Issue 1: I can ping the virtual servers from my own hardware and from the remote routers. However, I cannot load the web page admin nor can I SSH into the CUCM/CUC devices. Issue 2: BR1 gateway has a line protocol down for VLAN interfaces 130 and 240. I mentioned this on the OSL a few weeks prior. The basic issue is that those interfaces will never show an UP/UP state. I have reloaded the router, removed configs, compared with previous working configs, etc. Still no dice. I made sure that the interfaces were configured properly. In fact, the same configuration works on BR2 (different VLAN ID and IPs, off course). Below are some outputs to defend issue two: BR1-RTR#show int vlan 130 Jun 14 00:44:05.575: %SYS-5-CONFIG_I: Configured from console by console BR1-RTR#show int vlan 130 Vlan130 is up, line protocol is down Hardware is EtherSVI, address is 0017.9460.8d40 (bia 0017.9460.8d40) Internet address is 10.10.101.1/24 MTU 1500 bytes, BW 10 Kbit/sec, DLY 100 usec, reliability 255/255, txload 1/255, rxload 1/255 Encapsulation ARPA, loopback not set ARP type: ARPA, ARP Timeout 04:00:00 Last input never, output never, output hang never Last clearing of show interface counters never Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 0 Queueing strategy: fifo Output queue: 0/40 (size/max) 5 minute input rate 0 bits/sec, 0 packets/sec 5 minute output rate 0 bits/sec, 0 packets/sec 0 packets input, 0 bytes, 0 no buffer Received 0 broadcasts, 0 runts, 0 giants, 0 throttles 0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored 0 packets output, 0 bytes, 0 underruns 0 output errors, 0 interface resets 0 unknown protocol drops 0 output buffer failures, 0 output buffers swapped out BR1-RTR#show ip int vlan 130 Vlan130 is up, line protocol is down Internet address is 10.10.101.1/24 Broadcast address is 255.255.255.255 Address determined by setup command MTU is 1500 bytes Helper address is not set Directed broadcast forwarding is disabled Multicast reserved groups joined: 224.0.0.5 Outgoing access list is not set Inbound access list is not set Proxy ARP is enabled Local Proxy ARP is disabled Security level is default Split horizon is enabled ICMP redirects are always sent ICMP unreachables are always sent ICMP mask replies are never sent IP fast switching is enabled IP fast switching on the same interface is disabled IP Flow switching is disabled IP CEF switching is enabled IP CEF switching turbo vector IP Null turbo vector IP multicast fast switching is enabled IP multicast distributed fast switching is disabled IP route-cache flags are Fast, CEF Router Discovery is disabled IP output packet accounting is disabled IP access violation accounting is disabled TCP/IP header compression is disabled RTP/IP header compression is disabled Policy routing is disabled Network address translation is disabled BGP Policy Mapping is disabled WCCP Redirect outbound is disabled WCCP Redirect inbound is disabled WCCP Redirect exclude is disabled BR1-RTR#show vlan-switch brief VLAN Name StatusPorts - --- 1default activeFa1/0, Fa1/1, Fa1/2, Fa1/3 Fa1/4, Fa1/5, Fa1/6, Fa1/7 Fa1/8, Fa1/9, Fa1/10, Fa1/11 Fa1/12, Fa1/13, Fa1/14, Fa1/15 130 DATA active 240 PHONES activeFa1/1, Fa1/2, Fa1/3, Fa1/4 Fa1/5, Fa1/6, Fa1/7, Fa1/8 Fa1/9, Fa1/10, Fa1/11, Fa1/12 Fa1/13, Fa1/14, Fa1/15 1002 fddi-default act/unsup 1003 token-ring-default act/unsup 1004 fddinet-default act/unsup 1005 trnet-defaultact/unsup interface FastEthernet1/1 switchport trunk native vlan 130 switchport mode trunk switchport voice vlan 240 spanning-tree portfast ! interface FastEthernet1/2 switchport trunk native vlan 130 switchport mode trunk switchport voice vlan 240 spanning-tree portfast ! interface Vlan130 ip address 10.10.101.1 255.255.255.0 ! interface Vlan240 ip address 10.10.201.1 255.255.255.0 ip helper-address 10.10.200.3 -- *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab
Re: [OSL | CCIE_Voice] problem LAB 11 VOL1 CUE-CME license file- working on vrack
Could you service-engine interface be shut down? *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 6/13/2010 7:13 PM, amr gaber wrote: I try to upload CUE -CME license bu the command below software install clean url ftp://10.10.210.5/cue-vm-license_12mbx_cme_7.0.1.pkg username cisco password cisco please advise as soon as possible Logging se-10-10-202-2# $e_12mbx_cme_7.0.1.pkg username cisco password cisco WARNING:: This command will install the necessary software to WARNING:: complete a clean install. It is recommended that a backup be done WARNING:: before installing software. Would you like to continue? [n]y Downloading ftp cue-vm-license_12mbx_cme_7.0.1.pkg Error: Download error Can not download cue-vm-license_12mbx_cme_7.0.1.pkg error code 0 : error type 'couldn't connect to host' se-10-10-202-2# ___ 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] Vol 2 Lab 7 - CUC SIP Integration
Anyone had any success with this? I am following the Proctor Guide and only get a fast busy. I also followed the Cisco document and have not been able to get a successful call over a SIP trunk into CUC. Ideas? This is what I have done in CUCM: - Create SIP Trunk Security Profile Accept Out-of-Dialog REFER Accept Unsolicited Notification Accept Header Replacement - Create SIP Profile - Create SIP Trunk Calling Search Space Redirecting Diversion Header Delivery - Inbound Redirecting Diversion Header Delivery - Outbound Rerouting Calling Search Space Out-of-Dialog Refer Calling Search Space SIP Profile - Create Route List - Create Route Group - Create Route Pattern On-Net - Create Voicemail Mail Pilot - Create Voicemail Profile - Create Application User Accept Out-of-Dialog REFER Accept Unsolicited Notification Accept Header Replacement I have also done the CUC-related steps but don't have them typed out. -- *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 2 Lab 7 - CUC SIP Integration
I figured it out. There were hunt pilot remnants of a SCCP integration existent on the VM image. *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 6/11/2010 6:42 PM, Matthew Berry wrote: Anyone had any success with this? I am following the Proctor Guide and only get a fast busy. I also followed the Cisco document and have not been able to get a successful call over a SIP trunk into CUC. Ideas? This is what I have done in CUCM: - Create SIP Trunk Security Profile Accept Out-of-Dialog REFER Accept Unsolicited Notification Accept Header Replacement - Create SIP Profile - Create SIP Trunk Calling Search Space Redirecting Diversion Header Delivery - Inbound Redirecting Diversion Header Delivery - Outbound Rerouting Calling Search Space Out-of-Dialog Refer Calling Search Space SIP Profile - Create Route List - Create Route Group - Create Route Pattern On-Net - Create Voicemail Mail Pilot - Create Voicemail Profile - Create Application User Accept Out-of-Dialog REFER Accept Unsolicited Notification Accept Header Replacement I have also done the CUC-related steps but don't have them typed out. -- *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Sip Dial rule
I wouldn't think it would be required unless explicitly stated to do so. If your phones go into SRST-mode, they will still use the dial rule configured under CUCM since it is downloaded to the phone itself. *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 6/9/2010 9:55 PM, So Gwaai wrote: According to the CUCM system guide, the sip dial rule for 7965 is optional since this phone type run KPML. And under srst mode, sip phone use the dial rule which received from cucm. I've just want to know that whether we need to config the sip dial-rule for the SRST? ___ 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] FRTS and MLP over a Serial with Sub-Interfaces
Quick question. In the lab, if the HQ site is setup with two sub-interfaces that connect to BR1 and BR2 (i.e. meaning, they're both running off the same interface), how would you configure MLP for one site and FRF.12 for another site? According to my understanding, MLP will require that frame-relay traffic-shaping is enabled on the serial interface. However, this would botch up your FRF.12 configuration on the other sub-interface. QoS is a weak area for me so I might be missing something obvious in this question. However, it came up so I thought I would ask. Thanks -- *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] FRTS and MLP over a Serial with Sub-Interfaces
So FRTS for both sides is allowed. However, FRF.12 and MLP can only exist on separate physical interfaces? *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 6/10/2010 9:50 AM, kerboute kerboute wrote: The only way is to have separate interfaces, however you can use FRTS for both sites. On 06/10/2010 03:35 PM, Matthew Berry wrote: Quick question. In the lab, if the HQ site is setup with two sub-interfaces that connect to BR1 and BR2 (i.e. meaning, they're both running off the same interface), how would you configure MLP for one site and FRF.12 for another site? According to my understanding, MLP will require that frame-relay traffic-shaping is enabled on the serial interface. However, this would botch up your FRF.12 configuration on the other sub-interface. QoS is a weak area for me so I might be missing something obvious in this question. However, it came up so I thought I would ask. Thanks -- *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ For more information regarding industry leading CCIE Lab training, please visitwww.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Setting number plan indicator on the dial peer without a translation rule
Reading the Implementing Cisco Voice Gateways and Gatekeepers student guide, page 290. They cite another way to set numbering plan on a dial peer. Here is their example: dial-peer voice 100 pots *numbering-type national* destination-pattern 91408... prefix 1408 port 1/0:23 Has anyone tried this before? This might be a way to avoid (if needed) setting the type via a translation-rule/profile. Thoughts? -- *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] clock summer-time
Is it necessary to define a start/stop for the clock summer-time recurring command? I have been entering this as a general practice for all my exercises. However, I'm not sure if it's required to enter a start/stop time. Comments? -- *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 52, Issue 40 CUCM Subscriber as TFTP
Thanks for responding to this! *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 6/6/2010 5:02 PM, Beck, Ken wrote: That statement is for call processing only. TFTP and DHCP are excluded from those duties. Regards, Ken -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Sunday, June 06, 2010 12:49 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 52, Issue 40 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: VPIM error on CUE (554 Bad Sender's System) [Solved] (Pavan K) 2. How to send a secure message in Unity Connection ? (Pavan K) 3. CUCM Subscriber as TFTP (Matthew Berry) -- Message: 1 Date: Sun, 6 Jun 2010 13:04:08 -0500 From: Pavan Kpav.c...@gmail.com To: osl oslccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] VPIM error on CUE (554 Bad Sender's System) [Solved] Message-ID: aanlktimuxowggn4ajglpvjdccyr-kuoppbgr7txe8...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Had to change domain name on Unity connection under SMTP settings and reboot the box. Restarting the Conversation Manager service (as instructed by the GUI) didn't make any difference. -Pavan On Sat, Jun 5, 2010 at 7:41 PM, Pavan Kpav.c...@gmail.com wrote: Trying VPIM Sending messages from CUE to UnityConnection works perfectly. Messages from UnityConnection to CUE get an error message and generate a NDR (non-delivery receipt) Looking through the SMTP traces, i see a 554 error. (Screenshot attached). Anybody seen this before ? -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MVA on SIPGW
I don't think there are any plans. However, you can receive the call on an MGCP gateway and hairpin it through an internal H323 gateway if you so choose. Matthew Berry **Sent from my iPhone** Skype/Twitter: ciscovoiceguru Google Voice: +1 612 424 5044 On Jun 7, 2010, at 7:39 AM, Rogers Ochieng r.ochi...@mfient.com wrote: In Volume 1 walkthrough, Vik mention that MVA is no supported with SIP or MGCP in the current version of IOS in the Lab. Don’t know if it’s supported in new versions From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice- boun...@onlinestudylist.com] On Behalf Of Pavan K Sent: Tuesday, June 01, 2010 1:37 AM To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MVA on SIPGW When using SIPGW and trying to transfer a call, The INVITE with the diversion header reaches CCM but is getting blocked in there due to a Top level domain mismatch. Wondering if anybody got it to work ? -Pavan On Mon, May 31, 2010 at 1:55 PM, Pavan K pav.c...@gmail.com wrote: Has any body tried this ? -- - Pavan -- - Pavan ___ 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] VLAN interfaces down
Just an update:
Yesterday, I dropped the same configuration into my 3750 switch and
the VLANs showed an up state. I have not searched the bug toolkit yet
(camping this week), but the bahavior is strange.
Matthew Berry
**Sent from my iPhone**
Skype/Twitter: ciscovoiceguru
Google Voice: +1 612 424 5044
On Jun 4, 2010, at 8:40 AM, Patrick Fischer myciscov...@gmail.com
wrote:
Hi Matthew
Did you try the no autostate command within the vlan interface?
This forces the VLAN interface to be up even if there is no active
port configured to use it.
Regards
Patrick
Maybe you can check
2010/6/4 bkvalent...@gmail.com bkvalent...@gmail.com
Also if the vlans are disabled. Make sure the vlans are active.
Sent from my Verizon Wireless Phone
- Reply message -
From: Amy Ryan ar...@ipexpert.com
Date: Fri, Jun 4, 2010 7:33 am
Subject: [OSL | CCIE_Voice] VLAN interfaces down
To: Angel Perez gorr...@hotmail.com, amccar...@cciequest.com, ciscovoiceg...@gmail.com
Cc: osl osl ccie_voice@onlinestudylist.com
Matthew,
Did you do a ³sh cdp neighber² and verify what port the phone was pl
ugged
into? I did notice below that you are showing the configuration for
int
fa1/1 and the Proctor Lab Racks usually have the phone at BR1
plugged into
interface fa1/0 which is also not shown in your output as being set
up. If
the phone was plugged into interface fa1/0 and was not configured
properly
or shut, you would have this behavior.
If the vlan.dat file was not present in flash and the VLAN¹s were co
nfigured
as you have shown below, then they would not show up when you do a ³
sh
vlan-s bri², but they would show up in a ³sh ip int bri². See
ScreenShots:
BR1-RTR#sh vlan-s bri (with vlan.dat deleted from flash)
VLAN Name StatusPorts
-
---
1default activeFa1/1, Fa1/2, Fa1/3,
Fa1/4,
Fa1/5
Fa1/6, Fa1/7, Fa1/8,
Fa1/9,
Fa1/10
Fa1/11, Fa1/12, Fa1/13,
Fa1/14, Fa1/15
1002 fddi-default act/unsup
1003 token-ring-default act/unsup
1004 fddinet-default act/unsup
1005 trnet-defaultact/unsup
!
!
BR1-RTR#sh ip int bri
Interface IP-Address OK? Method Status
Protocol
...
Vlan13010.10.101.1 YES NVRAM up
down
Vlan24010.10.201.1 YES NVRAM up
down
I will be around on Sunday. If you are able to recreate this then,
contact
me and I can hop into your session and look around.
Thank you,
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
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From: Angel Perez gorr...@hotmail.com
Date: Fri, 4 Jun 2010 08:03:41 +
To: amccar...@cciequest.com, ciscovoiceg...@gmail.com
Cc: osl osl ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] VLAN interfaces down
Hi:
If the vlan.dat file is deleted you will get this result
Make sure that the vlan exists and also that it is active:
vlan 130
create
name data
status active
vlan 240
create
name voice
status active
hth
Date: Thu, 3 Jun 2010 19:08:22 -0400
From: amccar...@cciequest.com
To: ciscovoiceg...@gmail.com
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] VLAN interfaces down
Hey Bro,
I ran into an issue similar to that before but mine was because
there
was no phone connected to the port.
Antonio
Quoting Matthew Berry ciscovoiceg...@gmail.com:
I see this issue from time to time. The VLAN interfaces on my
BR1-RTR
show a state of up, but line protocol is down.
I made sure that there are ports with the vlans configured. I
reload
the router. I also made sure the vlans were in existence.
At this point, my BR1-RTR is useless until I get this working.
Any ideas?
interface Vlan130
ip address 10.10.101.1 255.255.255.0
!
interface Vlan240
ip address 10.10.201.1 255.255.255.0
...
interface FastEthernet1/1
switchport trunk native vlan 130
switchport mode trunk
switchport voice vlan 240
BR1-RTR#show vlan-switch br
VLAN Name Status Ports
-
---
...
130
[OSL | CCIE_Voice] CUCM Subscriber as TFTP
I noticed that all the IPX labs state: For any tasks requiring redundancy or resiliency in utilizing the CUCMs, ensure that the Subscriber server is the primary for all functions listed. That said, is anyone using the Subscriber as the TFTP server, with the Publisher as its secondary. So far, my observations would say no. In real world, I almost always use the Publisher as primary TFTP and set the Subscribers to secondary. What is everyone's take on this? -- *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUCM Subscriber as TFTP
For lab preparation, are you always using the Sub as primary TFTP? *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 6/6/2010 3:02 PM, Ashar Siddiqui wrote: I have come across many customers who have Sub as Pri TFTP while Pub as secondary...nothing unusual there.. Ash Matthew Berry wrote: I noticed that all the IPX labs state: For any tasks requiring redundancy or resiliency in utilizing the CUCMs, ensure that the Subscriber server is the primary for all functions listed. That said, is anyone using the Subscriber as the TFTP server, with the Publisher as its secondary. So far, my observations would say no. In real world, I almost always use the Publisher as primary TFTP and set the Subscribers to secondary. What is everyone's take on this? -- *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ For more information regarding industry leading CCIE Lab training, please visitwww.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Volume 2 Lab 1 4.2 4.3
Are you using Proctor or your own lab setup? *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 6/5/2010 6:37 PM, Daniel Zeiger Berlinski wrote: Hello there I have completed the gatekeeper routing section of this lab and while testing I noticed that everytime I ring any BR2 phones from either HQ or BR1 using g711ulaw from CUBE to CME the call drops after 1 minute apprx. Looking further I noticed that all WAN bandwidth I have, is taken to the point that OSPF adjacency is lost. (in the case of my devices I have 128Kbps for these Frame tails because of hardware limitations of my lab) Well, show gatekeeper call displays exactly how the question mandates and supplementary services such as hold work as well but just for apprx 1 minute for the reasons I mentioned before. If I hop on my Frame switch I see the bandwidth consumption going higher and higher as time elapses. I'm running this setup with 2801 routers and 12.4(20)T2 advanced enterprise code. In essence what I'm seeing here is g711/g729 calls are consuming bandwidth until no more WAN bandwidth is available. I am starting to suspect of this being bug related? I'm not able to see the reason behind such behaviour and would be greatful if someone could help. ___ 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] vol 2 lab 1: gatekeeper and H.323 GW on the same router
A good rule of thumb is this: Only use the bind srcaddr when configuring an H.323 gateway to register with CUCM. If you are standing up a CME gateway, you enter the h323-gateway voip interface instead. *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 6/4/2010 7:53 PM, Daniel Zeiger Berlinski wrote: Hello Kobel I'm doing a few tests here and it seems that the difference between those 2 commands relates to which interface you are sourcing your RAS and H225 packets. For instance if you remove h323-gateway voip bind srcaddr from the interface which contains the IP address of the CUCM configured H323 gateway I believe you are going to break your incoming HQ local PSTN calls. Let me know if you found any different result. Thanks On Sat, Jun 5, 2010 at 3:37 AM, kobel findko...@gmail.com mailto:findko...@gmail.com wrote: bingo! indeed, this solved the issue. This command obviously binds all H.323 signaling to specific interface (loopback in my case). So this explains why the incoming calls were associated with the GK-controlled trunk (GK configured on loopback). After removing this command, the source address is bound with voice interface IP address, which is ok. Another lesson learned - for calls routed via GK, the SETUP which CUCM receives contains the source signaling IP address of the GK (despite the fact it's sent directly by the remote GW). This is why CUCM needs to sent ARQ to GK. Only ACF contains the signaling IP address of the remote GW. Now it makes perfect sense, but I've never though about it. But still, one thing is not clear for me. What's the difference between: * h323-gateway voip interface * h323-gateway voip bind srcaddr 10.225.100.254 The command reference is not very clear on this. Thanks for your input! On Fri, Jun 4, 2010 at 5:08 PM, Pavan pav.c...@gmail.com mailto:pav.c...@gmail.com wrote: in your existing config, remove the h323gw bind source interface command ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://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] VLAN interfaces down
I see this issue from time to time. The VLAN interfaces on my BR1-RTR show a state of up, but line protocol is down. I made sure that there are ports with the vlans configured. I reload the router. I also made sure the vlans were in existence. At this point, my BR1-RTR is useless until I get this working. Any ideas? interface Vlan130 ip address 10.10.101.1 255.255.255.0 ! interface Vlan240 ip address 10.10.201.1 255.255.255.0 ... interface FastEthernet1/1 switchport trunk native vlan 130 switchport mode trunk switchport voice vlan 240 BR1-RTR#show vlan-switch br VLAN Name StatusPorts - --- ... 130 DATA activeFa1/1, Fa1/15 240 PHONES activeFa1/1, Fa1/2, Fa1/3, Fa1/4 Fa1/5, Fa1/6, Fa1/7, Fa1/8 Fa1/9, Fa1/10, Fa1/11, Fa1/12 Fa1/13, Fa1/14, Fa1/15 BR1-RTR#show ip int bri Interface IP-Address OK? Method StatusProtocol ... Vlan13010.10.101.1 YES manual updown Vlan24010.10.201.1 YES NVRAM updown -- *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 52, Issue 15
BR1-RTR#show ip int bri Interface IP-Address OK? Method StatusProtocol FastEthernet0/0unassigned YES NVRAM administratively down down FastEthernet0/1unassigned YES NVRAM administratively down down Serial0/0/1:0 unassigned YES NVRAM upup Serial0/0/1:0.110.10.111.2 YES NVRAM upup FastEthernet1/0unassigned YES unset upup FastEthernet1/1unassigned YES unset updown FastEthernet1/2unassigned YES unset updown FastEthernet1/3unassigned YES unset updown FastEthernet1/4unassigned YES unset updown FastEthernet1/5unassigned YES unset updown FastEthernet1/6unassigned YES unset updown FastEthernet1/7unassigned YES unset updown FastEthernet1/8unassigned YES unset updown FastEthernet1/9unassigned YES unset updown FastEthernet1/10 unassigned YES unset updown FastEthernet1/11 unassigned YES unset updown FastEthernet1/12 unassigned YES unset updown FastEthernet1/13 unassigned YES unset updown FastEthernet1/14 unassigned YES unset updown FastEthernet1/15 unassigned YES unset updown Vlan1 unassigned YES NVRAM upup Vlan13010.10.101.1 YES NVRAM updown Vlan24010.10.201.1 YES NVRAM updown BR1-RTR#show cdp neighbors detail - Device ID: SEP001794DFFBE0 Entry address(es): IP address: 10.10.201.30 Platform: Cisco IP Phone 7960, Capabilities: Host Interface: FastEthernet1/0, Port ID (outgoing port): Port 1 Holdtime : 120 sec Version : P00308000900 advertisement version: 2 Duplex: full Power drawn: 6.300 Watts *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 6/3/2010 4:12 PM, Beck, Ken wrote: What port is your phone connected to? Usually the phone is in f1/0 Send a show cdp nei det Show ip int brief and include the fastethernet ports. Regards, Ken -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Thursday, June 03, 2010 2:05 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 52, Issue 15 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. VLAN interfaces down (Matthew Berry) -- Message: 1 Date: Thu, 03 Jun 2010 16:05:21 -0500 From: Matthew Berryciscovoiceg...@gmail.com To: OSL Groupccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] VLAN interfaces down Message-ID:4c081911.1080...@gmail.com Content-Type: text/plain; charset=iso-8859-1; Format=flowed I see this issue from time to time. The VLAN interfaces on my BR1-RTR show a state of up, but line protocol is down. I made sure that there are ports with the vlans configured. I reload the router. I also made sure the vlans were in existence. At this point, my BR1-RTR is useless until I get this working. Any ideas? interface Vlan130 ip address 10.10.101.1 255.255.255.0 ! interface Vlan240 ip address 10.10.201.1 255.255.255.0 ... interface FastEthernet1/1 switchport trunk native vlan 130 switchport mode trunk switchport voice vlan 240 BR1-RTR#show vlan-switch br VLAN Name StatusPorts - --- ... 130 DATA activeFa1/1, Fa1/15 240 PHONES activeFa1/1, Fa1/2, Fa1/3, Fa1/4 Fa1
Re: [OSL | CCIE_Voice] VLAN interfaces down
Yes. The interface was up. This is the weird part. I've encountered this issue in the past with Proctor Labs. Usually reloading the router will fix the problem. Not today. I actually have a family issue that requires me to cancel my rack session early. I won't be able to verify further today. I will repeat the same lab this Sunday and follow-up. *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 6/3/2010 4:29 PM, Mike Brooks wrote: Hey Matt, Is interface fastethernet 1/1 up ? I believe the VLAN interfaces 130 and 240 will show down if the vlans are not configured on an interface that is up. Mike On Thu, Jun 3, 2010 at 5:05 PM, Matthew Berry ciscovoiceg...@gmail.com mailto:ciscovoiceg...@gmail.com wrote: I see this issue from time to time. The VLAN interfaces on my BR1-RTR show a state of up, but line protocol is down. I made sure that there are ports with the vlans configured. I reload the router. I also made sure the vlans were in existence. At this point, my BR1-RTR is useless until I get this working. Any ideas? interface Vlan130 ip address 10.10.101.1 255.255.255.0 ! interface Vlan240 ip address 10.10.201.1 255.255.255.0 ... interface FastEthernet1/1 switchport trunk native vlan 130 switchport mode trunk switchport voice vlan 240 BR1-RTR#show vlan-switch br VLAN Name StatusPorts - --- ... 130 DATA activeFa1/1, Fa1/15 240 PHONES activeFa1/1, Fa1/2, Fa1/3, Fa1/4 Fa1/5, Fa1/6, Fa1/7, Fa1/8 Fa1/9, Fa1/10, Fa1/11, Fa1/12 Fa1/13, Fa1/14, Fa1/15 BR1-RTR#show ip int bri Interface IP-Address OK? Method StatusProtocol ... Vlan13010.10.101.1 YES manual updown Vlan24010.10.201.1 YES NVRAM updown -- *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com mailto:ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Vol 2 Lab 1 Question 4.6 - Time Period / Time Schedule in UCM
Question 4.6 asks me to restrict international dialing outside of normal business hours. In the question, there is no mention of a custom blocked greeting that must be played by the annunciator when a call is blocked. In the solutions guide to lab one, IPexpert actually configures three time periods: M-F Evening, M-F Morning, and Weekends. They are essentially going the route of configuring the OFF periods instead of the ON periods (i.e. times when international dialing should be allowed). They assign the Time Schedule (made up of the three time periods) to a partition called PT-TOD. They then create a TP in the PT-TOD partition with a Block this pattern - No error action. I am wondering if the same results could be created by putting the international dialing pattern in a partition (PT-INTL-ALLOW) and only setting that partition to be active from 7am - 7pm weekdays. It seems like it would be a lot less work and accomplish the same results. Thoughts? Can someone confirm my suspicions? -- *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Vol 2 Lab 1 Question 4.7 - progress_ind setup
In the Proctor Guide for Lab 1 Question 4.7 the following dial-peer is setup for emergency dialing from branch two: dial-peer voice 999 pots translation-profile outgoing 8digitANI destination-pattern 999 *progress_ind setup enable 3* port 0/0/0:15 forward-digits 3 clid strip name Why do they set progress_ind setup enable 3? I did not see any reason stated in the question to enter this command. From my research, that command is only used if the PSTN does not provide ringback tone or the IOS gateway does not cut through the audio to the originating device. Even so, I would think that you'd use the progress_ind alert enable 8 command if absolutely needed. Thoughts are appreciated. Sometimes I get thrown off by commands that appear in the solutions but aren't explained. If it's necessary, I'd like to know. So far, I've been able to satisfy the requirements of the question without entering the command. -- *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010I ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] MIPS per Conference Call and Transcoding Session
The solutions guide for lab one states that a conference session takes up 240 MIPS and a transcoder session takes 30-40 MIPS for a high complexity codec. Where do you find that information? I found the voice termination MIPS table in the CUCM SRND, but there was no mention of transcoding/conferencing MIPS usage. Help? -- *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] H323 Gateway - Called Party Number Type: Unknown
This is what I have found to be true. Anyone, feel free to correct me if I am wrong. If you set called/calling number types in CUCM, they will not be passed to the H.323 gateway. It can be misleading, though, because the IOS has an algorithm that detects the number type based on the numbers that are sent to the gateway. If seven digits are sent, not beginning in 9, the gateway will mark it as SUBSCRIBE. The same applies for NATIONAL and INTERNATIONAL. According to what I have been told, this algorithm is based on the isdn switch-type primary-ni which is used by carriers servicing the NANP. If you change this to PRIMARY-NET5, it won't apply because the numbering plan changes. If you strip the 9 before sending calls to the gateway, you probably haven't noticed. However, if you send the 9 to the gateway for SRST purposes, then you'll likely run into this. I cannot lab this to verify, but I am 80% of this. *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 5/28/2010 12:13 PM, Angel Perez wrote: Hi: Do you have any called party transformation in the gw called party transformation calling search space? hth Date: Fri, 28 May 2010 11:35:42 -0500 From: tamnhu...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] H323 Gateway - Called Party Number Type: Unknown Hi all, Not sure if someone already posted the issue below or not, but I could not find one on OSL, so I post it here. The problem I have is the H323 gateway outbound called party number Type always show Unknown, even though I set it to National in the UCM. However, my BR1 MGCP gateway shows correct Type: National. Here is the call flow: HQ phone -- dialling 16178632683 -- TP 9.1617XXX [Called Party Num Type = Nation] -- RP \+1[2-9]xx[2-9]xx -- rg-local-gw It doesn't make any different when I tried to set the Type at the RP or TP. Also, the Calling Party Num Type is Unknown as well, even though, the 5XXX Calling Party Xform Pattern set to National Any suggestions would be apppricated. Thanks, Tam May 28 16:36:24.854: ISDN Se0/2/0:23 Q931: TX - SETUP pd = 8 callref = 0x0090 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-PHN1' Calling Party Number i = 0x0081, '+12123945001' *Plan:Unknown, Type:Unknown *Called Party Number i = 0x80, '16178632683' *Plan:Unknown, Type:Unknown *May 28 16:36:24.878: ISDN Se0/2/0:23 Q931: RX - CALL_PROC pd = 8 callref = 0x8090 Channel ID i = 0xA98383 Exclusive, Channel 3 Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. https://signup.live.com/signup.aspx?id=60969 ___ 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] Accessing Unity Express Voicemail from outside
I doubt there would be any additional toll-fraud vulnerabilities by having a translation-rule forward to the internal voicemail number. The same vulnerabilities would apply if your voicemail pilot was in your DDI range. You can restrict outbound call from within Unity Connection by using PT/CSS on the ports or by limiting the allowable dialable digits in CUC. *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 5/28/2010 5:05 AM, Ashar Siddiqui wrote: Hi all, Another Unity express related query... Customer wanted to access their voicemail from outside. The Voicemail pilot was not in DDI range so I created a translation rule which is now converting one of their DDI number to Voicemail. My concern is that are there any possibilities of misusing this feature, I mean toll-fraud etc. If yes, then how can I restrict it. Any help would be much appreciated. Thanks Ash ___ 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] Vol2 Lab_1
Try removing the ip helper-address 10.10.210.11 from your interface. If you want multiple helper addresses, you can do them on a single line/command. *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 5/28/2010 1:46 PM, Gregory Bonton wrote: ip helper-address 10.10.210.11 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] qos question
Answers inline below: 1)With MLP you use as FR header 11 byte (i.e. Vol2 solution Labs 1-5, page 111 or Vol2 solutionLabs 1-5 page 559) but sometimes you use 9byte instead (i.e. Vol2 solution Labs 6-10 page 595 or in VOD DVD Task 10.3-10.4, time 13:43). In Vol2 solution Labs 1-5 page 877 it is 4 byte. Wich one is correct? 4,9 or 11??? (also SRND report 13 byte) /* My comments: The different byte counts are based on what type of layer two technology is being used. The QoS SRND allocates 12 bytes for PPP, 13 bytes for MLP, 4 bytes for Frame-Relay, and 8 bytes for FRF.12. Vik has done some calculations based on packet captures or some kind of magic and determined that MLP actually only takes 9 bytes, not 13 (Vik - Correct me if I'm wrong). In the end, your calculation will not be much different if you choose 9 bytes or 13 bytes. Maybe someone else could chime in here, but I think I remember reading that proctors allow for a 10% variance.*/ 2)Vol2 - Lab 3 – page 59 Question 5.2 , you ask “Signaling traffic emerging from phones..” and in Vol2 solution pag 557 you answer “mls qos srr-queue OUTPUT cos-map ..” , but, as far as I understand, traffic FROM phone is input queue on the switch and not output (output queue should be traffic from switch TO phone).. Can you please tell me if I am wrong and why? /*/*My comments: It all depends on perspective. If you're looking from the standpoint of what the router will be seeing, mapping values at either point will result in the same perspective. You have more options on the four egress queues. Perhaps that is why the guide shows editing the egress queue. In real life, you will rarely change the ingress queue settings.*/*/ 3)Vol2 solution Labs 6-10 page 73 and 74 question 7.1 – why you trust (match dscp under class-map)?.. because there is no qos configured on any switch, I would have not trusted dscp but, instead, matched protocol or access-group and then set dscp under policy-map.. I always thought to trust on router when qos is enabled on the switch but in the soltuion provided it does not work this way.. so, can you please tell me when to trust? /*My comments: CUCM will already mark RTP packets according to DSCP. I would generally trust what I am told to trust. If the question does not explicitly tell me what to trust/not-trust, I would just pick an option. My guess is that the scoring would be based on other components of your QoS configuration, not how you chose to classify the traffic. That said, however, it's important to make sure it's properly classified.*/ *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 5/27/2010 8:33 AM, gabriele lietti wrote: Hi, Hope you all are ok.. I have some questions about QoS.. hope someone can help me: 1) With MLP you use as FR header 11 byte (i.e. Vol2 solution Labs 1-5, page 111 or Vol2 solutionLabs 1-5 page 559) but sometimes you use 9byte instead (i.e. Vol2 solution Labs 6-10 page 595 or in VOD DVD Task 10.3-10.4, time 13:43). In Vol2 solution Labs 1-5 page 877 it is 4 byte. Wich one is correct? 4,9 or 11??? (also SRND report 13 byte) 2) Vol2 - Lab 3 – page 59 Question 5.2 , you ask “Signaling traffic emerging from phones..” and in Vol2 solution pag 557 you answer “mls qos srr-queue OUTPUT cos-map ..” , but, as far as I understand, traffic FROM phone is input queue on the switch and not output (output queue should be traffic from switch TO phone).. Can you please tell me if I am wrong and why? 3) Vol2 solution Labs 6-10 page 73 and 74 question 7.1 – why you trust (match dscp under class-map)?.. because there is no qos configured on any switch, I would have not trusted dscp but, instead, matched protocol or access-group and then set dscp under policy-map.. I always thought to trust on router when qos is enabled on the switch but in the soltuion provided it does not work this way.. so, can you please tell me when to trust? thanks in advance ___ 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] Vol 2 Lab 4 Section 3.1 - Mobile Connect Question
That did the trick. That's one thing I'll never forget. Thanks, Peter. *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 5/26/2010 1:49 AM, Peter Farkas wrote: Display ID of RDP's DN is missing. When shared line is created then only the Alerting Name is copied from the line. Go to the DN Configuration of 5002 and select the RDP from Associated Devices list and use Edit Line Appaearance button to modify. - Original Message - *From:* Matthew Berry mailto:ciscovoiceg...@gmail.com *To:* OSL Group mailto:ccie_voice@onlinestudylist.com *Sent:* Wednesday, May 26, 2010 3:11 AM *Subject:* [OSL | CCIE_Voice] Vol 2 Lab 4 Section 3.1 - Mobile Connect Question Fellow nerds, I am battling a single number reach (i.e. Mobile Connect) question on Lab 4. Question 3.1 says the call should appear to BR1 Phone 2 as if it is actually coming from HQ Phone 2 directly (Calling Name and Number). When I call in from the PSTN phone to BR1 Phone 2, the display on BR1 Phone 2 shows 5002. The calling number is represented just fine. However, I cannot get the calling nmae to be presented on the display. I have tinkered around with the partial/complete match and significant digits parameters under the mobility section of the Call Manager service parameters but nothing has changed. Any ideas? -- *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ 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] Accessing Unity Express Voicemail from outside
Sure. Sounds good. I apologize for not reading your email in its entirety. Although, I'm sure the same applies regardless of CUE/CUC. Happy labbing *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 5/28/2010 2:20 PM, Ashar Siddiqui wrote: Thanks Matt. It's actually Unity Express and the customer is using CME. I know there is an option where you can restrict transfer by using restriction tables. I later found that customer has allowed transfers to 9T pattern under tranfer-pattern (which could be everything) so I don't think there is any need of putting in restrictions. Ash Matthew Berry wrote: I doubt there would be any additional toll-fraud vulnerabilities by having a translation-rule forward to the internal voicemail number. The same vulnerabilities would apply if your voicemail pilot was in your DDI range. You can restrict outbound call from within Unity Connection by using PT/CSS on the ports or by limiting the allowable dialable digits in CUC. *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 5/28/2010 5:05 AM, Ashar Siddiqui wrote: Hi all, Another Unity express related query... Customer wanted to access their voicemail from outside. The Voicemail pilot was not in DDI range so I created a translation rule which is now converting one of their DDI number to Voicemail. My concern is that are there any possibilities of misusing this feature, I mean toll-fraud etc. If yes, then how can I restrict it. Any help would be much appreciated. Thanks Ash ___ For more information regarding industry leading CCIE Lab training, please visitwww.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visitwww.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] EM CUCME Problem
Did you create cnf-files and reload the phones? You can do a debug ip http events to see what's going on when you press the services button. *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 5/24/2010 8:02 AM, Patrick Fischer wrote: Hi Did you configure url authentication ... under telephony-service? Regards Patrick 2010/5/24 kerboute kerboute naoufal.kerbo...@cbi.ma mailto:naoufal.kerbo...@cbi.ma Hi guys, I have an issue with EM on CME BR2, I've created the logout profile and assign it to the br2 phone2 but when i press the button service I've got No services Configured, is there any restriction due to the firmware of IP phones ?? Any Idea? note: ip http server already configured Thank you ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://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] CTI Gateway Setting in CUPS
When configuring the CTI gateway parameter in CUPS, I have a TCP or TLS option for each CUCM server (Pub and Sub). I'm 95% certain that we will always choose TCP. However, I'm trying to figure out if we would select the host for Pub or Sub. If the lab requirements are that the Sub is primary for all operations, would we follow course and select the Sub TCP host? -- Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CTI Gateway Setting in CUPS
One more follow-up comment. I'm guessing that we'd configure the Sub TCP host as primary and Pub TCP host as secondary (displayed below). Does that sound right? Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Weird Behavior in Unity Connection
I am trying to import users into Unity Connection. I've done this many times before. However, this time I am getting the following error: Wednesday, May 26, 2010 12:06:53 AM EDT ERROR importing user (gwashington) with extension = 5002 : The credential minimum length check failed. Minimum length = 1 . Wednesday, May 26, 2010 12:06:54 AM EDT ERROR importing user (jadams) with extension = 1002 : The credential minimum length check failed. Minimum length = 1 . Funny thing is that I have set the correct credential/password length in CUCM under End User. I also made sure that the LDAP username was within the correct bounds. I also tried creating different user templates and authentication rules to use while importing users. I also rebooted the box. No dice. Has anyone seen this before? -- *Matthew Berry* /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/ *_Vitals:_* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *_Cert Stats:_* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Location based RSVP over dual Frame Relay Links
You are not allocating enough bandwidth for two G711 calls with RSVP. One at 96 (worst case) and one at 64. Matthew Berry **Sent from my iPhone** Skype/Twitter: ciscovoiceguru Google Voice: +1 612 424 5044 On May 22, 2010, at 8:48 AM, Graham Hopkins ghopk...@wolf-rock.co.uk wrote: Has anyone got this working/had problems etc. I have two links with 96k allocated per link but the second call (both G711) gets Not Enough Bandwidth. routing is load-sharing O E2 192.168.50.0/24 [110/20] via 10.10.10.10, 00:23:33, Serial1/0.3 [110/20] via 10.10.10.2, 00:23:33, Serial1/0.1 RSVP call agents are up and registered to CUCM. Any ideas ? interface Serial1/0.1 point-to-point bandwidth 384 ip address 10.10.10.1 255.255.255.252 frame-relay interface-dlci 101 ip rsvp bandwidth 96 end HQ-GW#sh run int s1/0.3 Building configuration... Current configuration : 155 bytes ! interface Serial1/0.3 point-to-point bandwidth 384 ip address 10.10.10.9 255.255.255.252 frame-relay interface-dlci 111 ip rsvp bandwidth 96 end HQ-GW# HQ-GW#sh ip rsvp interface interfacersvp allocated i/f max flow max sub max Se1/0ena 80K1158K1158K0 Se1/0.1 ena 80K96K 96K 0 Se1/0.3 ena 0 96K 96K 0 dspfarm profile 1 mtp codec pass-through codec g711ulaw rsvp maximum sessions software 8 associate application SCCP O E2 192.168.50.0/24 [110/20] via 10.10.10.10, 00:23:33, Serial1/0.3 [110/20] via 10.10.10.2, 00:23:33, Serial1/0.1 Graham Hopkins ___ 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] Location based RSVP over dual Frame Relay Links
Graham, According to my understanding, the 64 Kbps does not equal 24 Kbps for the call and 40 Kbps for setup. Instead, the RSVP reservation always calculates the incoming call at the worst-case scenario of 40 Kbps for a g.729 call. The remaining 24 Kbps is for call #2. I am not familiar with lab 5 so I can't speak to the load balanced links. Could you send your gateway configs and the debug ip RSVP messages? Happy labbing! Matthew Berry **Sent from my iPhone** Skype/Twitter: ciscovoiceguru Google Voice: +1 612 424 5044 On May 22, 2010, at 10:06 AM, Graham Hopkins ghopk...@wolf- rock.co.uk wrote: Matthew - the two interfaces S1/0.1 and S1/0.3 are parallel links to the same remote site 96 K is allocated on each of the two links, enough for one call per link. This is based on Vol 2 Lab 5 scenario, according to the proctor guide the first call should use S1/0.1 and the second S1/0.3 but I never get a call on the second link - even if the bandwidth is set to 500K ! The actual example in Vol2 Lab 5 was to allow 4 calls at G.729 and the solution allowed 64K per sub interface ( i.e. 24K plus 40K for call setup) however I could not get more than two calls between the sites in this instance Regards Graham Hopkins On 22 May 2010, at 15:45, Matthew Berry wrote: You are not allocating enough bandwidth for two G711 calls with RSVP. One at 96 (worst case) and one at 64. Matthew Berry **Sent from my iPhone** Skype/Twitter: ciscovoiceguru Google Voice: +1 612 424 5044 On May 22, 2010, at 8:48 AM, Graham Hopkins ghopk...@wolf- rock.co.uk wrote: Has anyone got this working/had problems etc. I have two links with 96k allocated per link but the second call (both G711) gets Not Enough Bandwidth. routing is load-sharing O E2 192.168.50.0/24 [110/20] via 10.10.10.10, 00:23:33, Serial1/0.3 [110/20] via 10.10.10.2, 00:23:33, Serial1/0.1 RSVP call agents are up and registered to CUCM. Any ideas ? interface Serial1/0.1 point-to-point bandwidth 384 ip address 10.10.10.1 255.255.255.252 frame-relay interface-dlci 101 ip rsvp bandwidth 96 end HQ-GW#sh run int s1/0.3 Building configuration... Current configuration : 155 bytes ! interface Serial1/0.3 point-to-point bandwidth 384 ip address 10.10.10.9 255.255.255.252 frame-relay interface-dlci 111 ip rsvp bandwidth 96 end HQ-GW# HQ-GW#sh ip rsvp interface interfacersvp allocated i/f max flow max sub max Se1/0ena 80K1158K1158K0 Se1/0.1 ena 80K96K 96K 0 Se1/0.3 ena 0 96K 96K 0 dspfarm profile 1 mtp codec pass-through codec g711ulaw rsvp maximum sessions software 8 associate application SCCP O E2 192.168.50.0/24 [110/20] via 10.10.10.10, 00:23:33, Serial1/0.3 [110/20] via 10.10.10.2, 00:23:33, Serial1/0.1 Graham Hopkins ___ 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] DHCP in CCM on Proctor lab
Can't say I've ever tried. You might be able to get away using the ip helper-address on your local interfaces. Matthew Berry **Sent from my iPhone** Skype/Twitter: ciscovoiceguru Google Voice: +1 612 424 5044 On May 22, 2010, at 2:14 PM, Erwan Erwan e_er...@yahoo.com wrote: hi, Is anybody know how to activate DHCP in proctolab CCM for our home IP phones ? What need to condig in VPN router and Switches ? tks ___ 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] CME: IP source Address [any-match] and [strict-match]
No. Matthew Berry **Sent from my iPhone** Skype/Twitter: ciscovoiceguru Google Voice: +1 612 424 5044 On May 21, 2010, at 4:26 AM, Mahdi Mohood forccievo...@yahoo.com wrote: Thank you for your reply. Do you mean I have to use this if I have more than one CME and I need to restrict the registration of the phones ? --- On Fri, 5/21/10, Matthew Berry ciscovoiceg...@gmail.com wrote: From: Matthew Berry ciscovoiceg...@gmail.com Subject: Re: [OSL | CCIE_Voice] CME: IP source Address [any-match] and [strict-match] To: ccie_voice@onlinestudylist.com Date: Friday, May 21, 2010, 4:19 AM If you have a router with three different VLANS (i.e. different subnets), you could restrict phones on subnets 2 and 3 from registering with the CME sourced from an IP on subnet 1. This would rarely be used, but might be useful to restrict devices from registering. Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 5/20/2010 9:35 PM, Mahdi Mohood wrote: Hi all I tried to read about the difference between the two commands [any-match and strict-match] but I did not find the exact answer. I understood that we are using this command to allow or deny the registration of phones. I found this in the archive of on line study: Use the *any-match* keyword to instruct the router to permit Cisco IP phone registration even when the IP server address used by the phone does not match the IP source address. This option can be used to allow registration of Cisco IP phones on different subnets or those with different default DHCP routers or different TFTP server addresses. Use the* strict-match *keyword to instruct the router to reject Cisco IP phone registration attempts if the IP server address used by the phone does not exactly match the source address. By dividing the Cisco IP phones into groups on different subnets and giving each group different DHCP default-router or TFTP server addresses, this option can be used to restrict the number of Cisco IP phones allowed to register. I could not understand how the IP phone will register with CME regardless of the IP address? and what is the relation between this and subnets and DHCP servers. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -Inline Attachment Follows- ___ 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] CUCME - SIP Call Pickup
Friderich - Do you have "ip http server" and "ip http path ..." configured? A.lso be aware that SIP call pickup is not covered in this version of the blueprint, so long as the version remains at CME 7.0 Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 5/17/2010 4:01 PM, Friderich Claude wrote: Hello, I have configured a CCME with IP Phones running the SIP Firmware. I wanted to test the call pickup feature with SIP phones… Below the configuration voice register global mode cme source-address 10.10.230.1 port 5060 max-dn 10 max-pool 5 load 7945 SIP45.8-4-2S load 7941 SIP41.8-4-2S authenticate register call-feature-uri pickup http://10.10.230.1/pickup call-feature-uri gpickup http://10.10.230.1/gpickup create profile sync 0002236318330437 ! voice register dn 1 number 3001 pickup-call any-group pickup-group 1 name BR1 Phone 1 ! voice register dn 2 number 3002 pickup-group 1 name BR2 Phone 2 ! voice register dn 3 number 3003 pickup-group 1 shared-line max-calls 6 ! voice register dn 4 number 3004 pickup-group 1 Any idea why it doesn’t work ? When I press the GPickup Softkey followed by * I have the reorder tone … Claude Friderich PreSales Support NETCORE PSF S.A. 49 rue du Baerendall B.P.65 L-8201 Mamer Téléphone: 31 33 80-407 Fax: 31 33 80 8-407 GSM: 621 303 616 E-mail: cfrider...@netcore.lu -- This email was Anti Virus checked by Astaro Security Gateway. Disclaimer The information in this Internet e-mail is confidential and may be legally privileged. It is intended solely for the addressee. Access to this Internet e-mail by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. When addressed to our clients any opinions or advice contained in this e-mail are subject to the terms and conditions expressed in our governing terms of business. ___ 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
