Re: [OSL | CCIE_Voice] Scheduling Lab Exam within 30 Days
Ken, I have booked with only like 2 weeks left before the exam date. I'm pretty sure you have to pay the lab fees up front when you are in the 90 day period, therefore you can't be dropped due to no payment of fees. Jeff -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ken Wyan Sent: Thursday, August 25, 2011 1:50 AM To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Scheduling Lab Exam within 30 Days Hi, When trying to schdule a CCIE Lab within 90 days we get this message. You are booking a lab date within the 90 day payment window. Once you book this lab date, you are liable for the lab fee, and the lab date cannot be deleted or moved. Clicking below is your acknowledgement and agreement to pay the lab fee. Anybody booked exam less than 90 day period dropped due to non-payment of fees ? After dropping also does cisco ask for lab fee? Regards, Wyan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Scheduling Lab Exam within 30 Days
Ken, I have no experience in this scenario. I would just recommend if you aren't sure whether you're ready and/or can make it, then don't book. As Earl noted on another response and I did below, you will be charged immediately when you book in the 90 day window. I don't know what you mean by 1 day to make the payment. You won't reserve your seat for the lab if you don't pay. Hope this helps, Jeff -Original Message- From: Ken Wyan [mailto:kew...@gmail.com] Sent: Thursday, August 25, 2011 6:42 AM To: givemeccievoice2...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Scheduling Lab Exam within 30 Days Jeff, After booking the exam , did you have to pay immediately to confirm your seat? Is it like web-booking of airtickets which doesn't guarant the seat until credit card payment is done? Because in China / Japan Wire Transfer is the only payment option available which cannot be done immediately. From cisco website I got following You may book an exam for a date less than 90 days away, if you complete payment on the day you book the exam. It seems they provide 1 day for payment. If they don't receive within 1 day what will happen? Any experience? Wyan On Thu, Aug 25, 2011 at 6:45 PM, givemeccievoice2...@gmail.com wrote: Ken, I have booked with only like 2 weeks left before the exam date. I'm pretty sure you have to pay the lab fees up front when you are in the 90 day period, therefore you can't be dropped due to no payment of fees. Jeff -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ken Wyan Sent: Thursday, August 25, 2011 1:50 AM To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Scheduling Lab Exam within 30 Days Hi, When trying to schdule a CCIE Lab within 90 days we get this message. You are booking a lab date within the 90 day payment window. Once you book this lab date, you are liable for the lab fee, and the lab date cannot be deleted or moved. Clicking below is your acknowledgement and agreement to pay the lab fee. Anybody booked exam less than 90 day period dropped due to non-payment of fees ? After dropping also does cisco ask for lab fee? Regards, Wyan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] exam cancel
I tried to cancel within the 3 month period, and they refused. Other than an absolute emergency I doubt they will grant you the ability to reschedule. Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of wilson.sam...@usc-bt.com Sent: Monday, July 25, 2011 9:08 AM To: f.faraday...@gmail.com; malexand...@uol.com.br Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] exam cancel Hi, With Voice Labs available across almost all the locations, I would prefer to go by Viks Advice of not booking the lab till the last 2 weeks, because almost certainly every continent they have the lab available with in the span of a week or two. Even then, I am not sure if the CSCO would accept any cancelation or postponing without forfeiting the fees Kind Regards Wilson Samuel From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of donny f Sent: Sunday, July 24, 2011 11:58 PM To: Marcelo Alexandria Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] exam cancel hi mar, what kind of porblem that Cisco certification can accept ? is it only sickness and doctor letter ? tks On Sun, Jul 24, 2011 at 11:48 AM, Marcelo Alexandria malexand...@uol.com.br wrote: Yes Donny, you can..you need open a case in cisco certifications page and justify your problem like a Doctor letter. De: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] Em nome de donny f Enviada em: domingo, 24 de julho de 2011 13:24 Para: ccie_voice@onlinestudylist.com Assunto: [OSL | CCIE_Voice] exam cancel hi all, does anyone has experience when need to drop off /move lab date? I heard sick and emergency , Cisco will able to do without lose the money And what needed , how many days notice ? tks Nenhum vírus encontrado nessa mensagem recebida. Verificado por AVG - www.avgbrasil.com.br http://www.avgbrasil.com.br/ Versão: 9.0.901 / Banco de dados de vírus: 271.1.1/3785 - Data de Lançamento: 07/24/11 03:33:00 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] route group distribution algorithm
If they aren't specific in the lab, then it's not a requirement of the solution. I don't believe I've ever seen or heard of this in the lab, more of a written exam question, if anything. Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cristobal Priego Sent: Wednesday, June 22, 2011 8:59 AM To: CCIE STUDENT Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] route group distribution algorithm the reason why i am asking is because in the lab they're not that specific 2011/6/22 CCIE STUDENT cciefo...@hotmail.com You rarely even do it in the real world -Original Message- From: Randall Saborio ill2...@gmail.com Date: Wed, 22 Jun 2011 13:26:24 To: cristobalpri...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] route group distribution algorithm Can you name when on the lab would you ever configure two devices on same route group? I don't think this is ever done on the lab. Even if it did, its not a matter of preference, but a matter of matching the task requirements. On Tue, Jun 21, 2011 at 7:41 PM, Cristobal Priego cristobalpri...@gmail.com mailto:cristobalpri...@gmail.com wrote: guys, for the lab, whenever you configure your route groups which distribution algorithm is better circular or top down i use top down all the time, but i'd like to know your opinion on this thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://www.PlatinumPlacement.com -- Randall da ill Saborio CCIE Voice Wannabe #10054675811 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Documents in the Lab
Everything here - http://www.cisco.com/cisco/web/psa/default.html?mode=prodlevel0=278875240 Also, a couple of the SRNDs will be available on the desktop. This is common knowledge, not breaking my NDA here :) Jeff -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ravindra Lakpriya Sent: Monday, June 06, 2011 3:26 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Documents in the Lab Hi Guys, What are the documents available for us to use during the exam ? just a general question :) -- Ravindra Lakpriya ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] SNR -- RC
If your Remote Destination is 4087773434, your route pattern in css-snr would need to look like that, not \+.!. Unless you are asked for redundancy with your GWs or to use the Application Dial Rules specifically, the easiest way to meet the SNR requirement is a SNR partition with a Route Pattern that will point directly to the local GW for that phone. Also, the complete match service parameter has to do with incoming from PSTN and/or recognizing for MVA, not SNR. I think you were assuming the 10 digit match would allow the route pattern to be used, but that is not the case. Hope this helps, Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Sam Park Sent: Monday, May 23, 2011 7:37 AM To: Cristobal Priego Cc: ccie vo...@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SNR -- RC So in those questions, what wording triggers you to use Application Dial Rules? Why can you just make your RD = +14087773434 and make a specific RP =\+1! to go out the specific gw; like Randall has? thanks: Sam On Sun, May 22, 2011 at 11:25 AM, Cristobal Priego cristobalpri...@gmail.com wrote: Did you configure your application dial rule to append a +1 ? Enviado desde mi iPhone El May 21, 2011, a las 21:01, Randall Crumm rrcr...@yahoo.com escribió: HI, Working on SNR I have it configured just like the Proctor guide and when 1002 calls 5002 the mobile phone does not ring. Everything else works correctly. I've had this issue before my RD is 4087773434 RDP rerouting css is css-snr CUCM services is compete match/10 RP= \+!/pt-snr\rl-hq I don't see anything hitting the hq rtr in the q931 debug Any ideas? Thanks, Randall ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] SW Version for Lab
Here is the official list of HW and SW in the lab - https://learningnetwork.cisco.com/docs/DOC-5292 All 7.0, particularly 7.0.1 as others have mentioned below. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Amit Singh Sent: Friday, April 29, 2011 5:22 AM To: George Goglidze Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SW Version for Lab Why 7.1 ? Regards Sent from my iPad On 29/04/2011, at 8:59 PM, George Goglidze gogli...@gmail.com wrote: In CUCME study for 7.1 Sent from my iPad On 28 Apr 2011, at 22:43, Abel ... midga...@gmail.com wrote: Hi everyone, the following list is the recommended software to be use on lab, is ok use the same version under the major release or must be use the higher version under minor release for v7.x of each one? Any major software release which has been generally available for six months is eligible for testing in the CCIE Voice Lab Exam. oCisco Unified Communications Manager 7.0 oCisco Unified Communications Manager Express 7.0 oCisco Unified Contact Center Express 7.0 oCisco Unified Presence 7.0 oCisco Unity Connection 7.0 Thanks Abel Mateo CCIE R/S 28546 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] H323 gateway doesn't send the plus on outgoing PRI calls
Adam, This is correct. If you have a requirement to send the + you'll have to add at voice-port using a translation-rule. Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of adam compton Sent: Wednesday, March 23, 2011 5:19 PM To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] H323 gateway doesn't send the plus on outgoing PRI calls Well, I might have found my answer: http://www.voiceie.com/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=8;t=002862 Poster says that h323 gateways can't process the plus. Can anyone confirm? Adam Compton On Wed, Mar 23, 2011 at 8:15 PM, adam compton com...@gmail.com wrote: I'm using external calling number mask with full e164 number. When I call from an extension out a MGCP gateway, the call goes as expected with the +. If I dial out of an H323 gateway, the external number shows with no plus. Anybody ran into this before? I can add the plus with a translation-profile on the voice-port, but in the back of my head, I feel i must be doing something wrong on the Call Manager side of things. Adam Compton ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Calling and Called Party Number Type
We technically arent allowed to answer your question about the lab. Dont stress out though, if the PSTN router wont accept something or is expecting something, its a safe bet that you will be told the information you need. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ccie Voice Sent: Thursday, March 03, 2011 10:52 AM To: CCIE Study Subject: Re: [OSL | CCIE_Voice] Calling and Called Party Number Type Thank you all for your reply, I just need to know if the PSTN router in the LAB will accept the call or no if it is not set to the proper value. If the PSTN router will not accept the call then it is OK I can play with these values and solve the problem. But the problem if the PSTN router accepts all calls based on called party number and later on the proctor will check if you set the values correctly or not. for me what I understood before is the way that Roger sent. (thank you Roger) Regards, _ From: Roger Källberg roger.kallb...@cygate.se To: Ccie Voice v.c...@yahoo.com; CCIE Study ccie_voice@onlinestudylist.com Sent: Thu, March 3, 2011 6:41:12 PM Subject: SV: [OSL | CCIE_Voice] Calling and Called Party Number Type Hi, You need to look at this from the originating endpoint and the outgoing gateway. For a more detailed explanation see my response in line with your mail. Sincerely Roger Källberg CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ _ Från: Ccie Voice [v.c...@yahoo.com] Skickat: den 3 mars 2011 02:49 Till: CCIE Study Ämne: [OSL | CCIE_Voice] Calling and Called Party Number Type Hi All, I am a little bit confused about how to set the value for Calling and Called Party Number Plan. let us say HQ Phone 1 Calls local Call in this case I think I have to set: Calling Party Number Type to: Subscriber. Called Party Number Type to: Subscriber. This is correct What about Long Distance: Calling Party Number Type to: Subscriber or National From the perspective of caller and VGW this is a call that came from a local site , aka it's subscriber Called Party Number Type to: National From the perspective of called and VGW this is a call goes to a remote phone, aka it's national it will be more complicated if we need to use TEHO, So if HQ Phone 1 calls BR1 Local PSTN number what I should set the values? Long Distance, using BR1 Router Calling Party Number Type to: Subscriber or National From the perspective of caller and VGW this is a call that came from a remote site , aka it's national Called Party Number Type to: National or Subscriber From the perspective of called and VGW this is a call goes to a local phone, aka it's subscriber Long Distance, backup for BR1 using HQ Router Calling Party Number Type to: Subscriber or National From the perspective of caller and VGW this is a call that came from a local site , aka it's subscriber Called Party Number Type to: National or Subscriber I am using BR1 Router From the perspective of called and VGW this is a call goes to a remote phone, aka it's national Regards, ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] h245 negotiation
It most likely has to do with your incoming dial-peer on BR2 CME. What do
you have configured for codec there? If you have nothing, then the codec
default is g729r8.
HTH,
Jeff
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of natan 2me
Sent: Tuesday, February 22, 2011 12:43 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] h245 negotiation
Hello. I am on BR2 CME with inbound call from CUBE (HQ RTR).
This is not question how do I make my call work... I just want to see why
g729AnnexA is being sent to me in INCOMING PDU from CUBE, while I have g711U
incoded into the OUTBOUND dial-peer on the CUBE (HQ RTR).
out PDU
capabilityTableEntryNumber 3
capability receiveAudioCapability : g711Ulaw64k : 20
in PDU
capabilityTableEntryNumber 6
capability receiveAudioCapability : g729AnnexA : 2
},
{
capabilityTableEntryNumber 3
capability receiveAudioCapability : g711Ulaw64k : 20
dial-peers from the HQ-RTR:
dial-peer voice 100 voip
incoming called-number .
dial-peer voice 200 voip
destination-pattern .T
session target ras
codec g711ulaw
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
Re: [OSL | CCIE_Voice] Barge/CBarge
Hi Ron, The point is that this is the expected behavior. If you don't want your screen cluttered you can use the privacy button to toggle privacy on/off in order to go from 2 displays to 1. To my knowledge there is no service parameter, feature, or anything else besides the privacy setting to effect this behavior. Jeff -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of rsmail...@solcon.nl Sent: Wednesday, February 09, 2011 10:33 AM To: matt...@ciscovoiceguru.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Barge/CBarge hmmm? is this a feature :) doesn't the phone have to do this itself ? because the cbarge is active and working acros the 3 phones. Ron While on the barged call, hit privacy and you'll see that second call go away. Matthew Berry, CCIE #26721 Email: matt...@ciscovoiceguru.com Twitter: http://twitter.com/CiscoVoiceGuru Blog: http://ciscovoiceguru.com On Feb 9, 2011, at 12:07 PM, rsmail...@solcon.nl wrote: hello matthew, privacy is off, because if it's on i can not barge in. take a look at the picture, you see what i mean with two call lines. Ron foto.JPG ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] I passed my Voice CCIE
Congrats Akash! Enjoy. -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Marko Milivojevic Sent: Friday, January 21, 2011 2:51 AM To: akash patel Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] I passed my Voice CCIE Well done! -- 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/ On Thu, Jan 20, 2011 at 17:45, akash patel akashapa...@yahoo.com wrote: 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 ___ 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] IPPA service url
You have the whole DocCD (http://www.cisco.com/cisco/web/psa/default.html?mode=prod http://www.cisco.com/cisco/web/psa/default.html?mode=prodlevel0=278875240 level0=278875240) and some of the SRNDs (including the Enterprise QoS SRND) available on the desktop. It's very much open book, but unfortunately you won't have the time to look for answers. Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of George Goglidze Sent: Friday, January 21, 2011 2:41 AM To: linuxboss.9 Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPPA service url Hi all, I was wondering if anyone can say what documentation we have exaclty in a lab. Is it only CUCM SRND? Or do we get UCCX Admin Guide, UC Admin Guide, Cisco IOS 12.4T Documentation (at least voice section) and all the other config guides available through cisco web page? Thanks, On Fri, Jan 21, 2011 at 4:29 AM, linuxboss.9 linuxbos...@gmail.com wrote: Easy way: EM, IPPM, IPMA all these service URLs are in CUCM SRND ( Which I believe is provided in lab) For UCCX like Jeff said: Doc CD Voice and Unified Communications Customer Collaboration Cisco United Contact Center Products Cisco Unified Contact Center Express Configuration Guides Configuration Examples and Tech Notes Configure a One Button Login for IP Phone Agents On Thu, Jan 20, 2011 at 9:38 AM, Randall Crumm randall.cr...@flextronics.com wrote: HI, Look at the CAD(Cisco Agent Desktop) installation guide page 53. HTH Randall From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mritunjay Kumar Sent: Thursday, January 20, 2011 6:19 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] IPPA service url Hi All, Where i can find service url for IPPA. I did not find in cucm and ccx help page ? any doc which is accessible in lab ? regards, Mritunjay Legal Disclaimer: The information contained in this message may be privileged and confidential. It is intended to be read only by the individual or entity to whom it is addressed or by their designee. If the reader of this message is not the intended recipient, you are on notice that any distribution of this message, in any form, is strictly prohibited. If you have received this message in error, please immediately notify the sender and delete or destroy any copy of this message ___ 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] IPPA service url
Doc CD Voice and Unified Communications Customer Collaboration Cisco United Contact Center Products Cisco Unified Contact Center Express Configuration Guides Configuration Examples and Tech Notes Configure a One Button Login for IP Phone Agents Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mritunjay Kumar Sent: Thursday, January 20, 2011 6:19 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] IPPA service url Hi All, Where i can find service url for IPPA. I did not find in cucm and ccx help page ? any doc which is accessible in lab ? regards, Mritunjay ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Presence btw CUCM and CME
Hi Mritunjay, Just to clarify what I meant from my notes. In your scenario below, the SCCP phone on CME should be able to monitor the CUCM SCCP, however the CUCM SCCP phone will not be able to monitor the presence of the CME SCCP phone. If you have a CUCM SIP Phone, then in 7.0.1 you should be able to monitor the CME phones. Hope this helps clarify, Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mritunjay Kumar Sent: Tuesday, January 18, 2011 2:03 AM To: Roger Källberg Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Presence btw CUCM and CME Hi Roger , both cucm and cme phone are sccp phones. Regards, Mritunjay 2011/1/18 Roger Källberg roger.kallb...@cygate.se The problem that was stated out in the to the previous tread linked to by kobel is for SCCP phones on CME. According to that you will only be able to get precense on SIP phones in CME Snipplet taken from that tread, I had written notes that Vik said CUCM SCCP Phone can't monitor CME phones in 7.0, but SIP should work fine. What kind of phones do you have on CME, SIP or SCCP? Roger Källberg CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ _ Från: Mritunjay Kumar [mjs...@gmail.com] Skickat: den 18 januari 2011 10:25 Till: Roger Källberg; Miron Kobelski Kopia: ccie_voice@onlinestudylist.com Ämne: Re: [OSL | CCIE_Voice] Presence btw CUCM and CME Hi Miron thanks for sharing this. i can conclude from this link that presence does not for following phone1-CUCM--SIP -CME-phone2 if phone1 is sccp then it cannot see the status of phone2. is this correct ? does any doc tell this ? hi Roger , I put the RP in partition which is accessible by phone's subscribe CSS but did not help :( Regards, Mritunjay 2011/1/18 Roger Källberg roger.kallb...@cygate.se Do you have subscribe CSS assigned to both SIP trunk and phone devices? Also try to add the RP to a partition that is seen in the subscribe CSS, should't make any difference, but I personaly prefer not to use the none PT. Reason is that you have less control of what a specific CSS should be able to see, but that's just my preference. Sincerely Roger Källberg CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ _ Från: Mritunjay Kumar [mjs...@gmail.com] Skickat: den 18 januari 2011 09:35 Till: Roger Källberg Kopia: ccie_voice@onlinestudylist.com Ämne: Re: [OSL | CCIE_Voice] Presence btw CUCM and CME Hi All, thank for reply. here is the answer allow watch is configured under ephone-dn set the CUP Publish trunk to respective trunk but did not help. i guess this is only for CUPS subscribe CSS is set to access the phone dn. Regards, Mritunjay 2011/1/18 Roger Källberg roger.kallb...@cygate.se Hi Mritunjay, In your config snipplet I don't see any allow watch on your ephone-dn's. See this url, http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/command/reference/cme_a1 ht.html#wp1016553 https://webmail.cygate.se/en/US/docs/voice_ip_comm/cucme/command/reference/ ,DanaInfo=.awxyCgnyjwImzy+cme_a1ht.html#wp1016553 Sincerely Roger Källberg CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ _ Från: Mritunjay Kumar [mjs...@gmail.com] Skickat: den 18 januari 2011 06:16 Till: ccie_voice@onlinestudylist.com Ämne: [OSL | CCIE_Voice] Presence btw CUCM and CME Hi All, CME phone is able to watch the status of CUCM phone but reverse is not working. config on CME presence presence call-list server 14.160.108.20 // .20 is sub max-subscription 120 watcher all allow subscribe presence enable sip-ua presence enable ephone 2 device-security-mode none mac-address 0022.9059.843D ephone-template 1 presence call-list blf-speed-dial 2 2001 label 2001 dial-peer voice 321 voip destination-pattern 2001 session protocol sipv2 session target ipv4:14.160.108.20 incoming called-number . on cucm phone1 (2001) has blf-speed dial to 4000 (CME phone) Route pattern 4XXX (Null partition )to SIP trunk (Towards CME) On SIP trunk , security profile is applied , valid check box ie accept presence sub and accept unsolicited are enabled. DN and trunk are in same presence group. Subscribe CSS can access DN 2001. when 2001 goes off hook , status can be seen on CME phone but other way , its not working. CUCM phone can make a call to CME phone by pressing BLF Speed dial Any missing config here ? Regards, Mritunjay ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Called # manupulation under SRST circumstance
Hi Shingei, I misread your question. Please disregard for this scenario. Jeff From: givemeccievoice2...@gmail.com [mailto:givemeccievoice2...@gmail.com] Sent: Tuesday, January 18, 2011 8:30 AM To: 'ccie_voice@onlinestudylist.com' Subject: RE: [OSL | CCIE_Voice] Called # manupulation under SRST circumstance Hi Shingei, Do all of your manipulations on the gateway when doing H323. This will save time for SRST in the actual lab. Here is how I normally do it: RP 9[2-9]XX RL/RG - H323 GW No digit manipulation, no use calling party external phone number mask. Just send 4 digit extensions and number as dialed. On gateway, voice translation-rule 1 rule 1 /^5.$/ /555\0/ type any subscriber plan any isdn voice translation-rule 2 rule 1 /^9/ // type any subscriber plan any isdn voice translation-profile SUBSCRIBER translate called 1 translate calling 2 dial-peer voice 1 pots destination-pattern 9[2-9]..$ port 0/0/0:23 translation-profile out SUBSCRIBER The called # will show on the screen as 7 digits 777 and the calling # will be sent as 7 digits 555 (just using random numbers to illustrate my point) and both will have subscriber/isdn as type/plan. Now, when you fall back to SRST, all of your dialing will work and the correct manipulations will occur before sending out to the gateway. Avoid using num-exp and dial-plan pattern commands when possible. Also, make sure you have your incoming voip dial-peer to set the codec and dtmf, I assumed you have this in place and the digit manipulation was the only issue. Hope this helps, Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ShinGei Yong Sent: Tuesday, January 18, 2011 3:56 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Called # manupulation under SRST circumstance Hi, I've below setup, BR2 as a mgcp gw with fall-back configured. dial-peer voice 9001 pots description *** SRST TO US *** destination-pattern 9001[2-9]..[2-9].. port 0/0/0:23 prefix 001 ! num-exp 5... 90012123945... num-exp 1... 90016178631... ! When in fallback mode,BR2 users dialed the 4digit number to reach HQ, the number get expanded to matched the outbound POTS dp with no issue, the call can established successfully. Problem is the called# presented on ipphone,instead of 4digit, the screen display 90012123945001,this is not the desired result. I did configured no supplementary-service h225-notify cid-update and reloaded but the result remain. How do i achieve the 4digit display in this case? Thanks Shingei. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SNR Line status
What type of phone are you using? For the remote in use, this is the correct way for it to work. Does your line turn red when you are using the mobile phone? From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of KatGuru Sent: Thursday, January 13, 2011 11:16 AM To: ccie_voice@onlinestudylist.com; Roger Källberg Subject: Re: [OSL | CCIE_Voice] SNR Line status Well, i turned off the privacy on both device and rdp but still there is no status on the desk phone unless you press the line button when the call is in progress with remote destination, then you see In use remote in the phone display. Thank you for your help though!! --- On Thu, 1/13/11, Roger Källberg roger.kallb...@cygate.se wrote: From: Roger Källberg roger.kallb...@cygate.se Subject: SV: [OSL | CCIE_Voice] SNR Line status To: KatGuru gkr2...@yahoo.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Thursday, January 13, 2011, 5:04 PM You need to turn off privacy on the remote destination profile and possible also on the desk phone. Roger Källberg CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ _ Från: KatGuru [gkr2...@yahoo.com] Skickat: den 13 januari 2011 06:16 Till: ccie_voice@onlinestudylist.com Ämne: [OSL | CCIE_Voice] SNR Line status Folks, Can we display Remote in use status in the desk phone if the call is answered in the cell phone? assume the desk phone is configured for snr. If so can any one please explain. Any help will be appreciated. Thank you. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Em on CME
Refer to the Administration Guide available on Cisco.com -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of iptuse...@hotmail.co.uk Sent: Wednesday, January 12, 2011 8:51 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Em on CME Im trying test EM on the cme but have an issue with service key not showing through EM service. Can anyone have a step by step guide on through url and getting it assigned to the service key Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUE intergarion in CUCM
Associate those CTI RP to the Application User and restart CUE. That should do the trick assuming everything else is ok. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Joli-coeur Wouter Sent: Wednesday, January 12, 2011 12:31 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CUE intergarion in CUCM Hi , I am trying to integrate my CUE in CUCM however i cant get it to work. When i check the integration status i get the following message: JTAPI Subsystem is not registered with any Call Manager This is the config on the the router voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip interface Service-Engine1/0 ip unnumbered FastEthernet0/0.112 no shutdown service-module ip address 177.3.12.2 255.255.255.0 service-module ip default-gateway 177.3.12.1 ip route 177.3.12.2 255.255.255.255 Service-Engine1/0 This is the config on the CUE username SCPH1 create username admin create username SCPH1 phonenumber 4001 ccn application ciscomwiapplication aa description ciscomwiapplication enabled maxsessions 2 script setmwi.aef parameter CallControlGroupID 0 parameter strMWI_OFF_DN 1998 parameter strMWI_ON_DN 1999 end application ccn application voicemail aa description voicemail enabled maxsessions 4 script voicebrowser.aef parameter logoutUri http://localhost/voicemail/vxmlscripts/mbxLogout.jsp; parameter uri http://localhost/voicemail/vxmlscripts/login.vxml; end application ccn subsystem jtapi ctiport 4221 4222 4223 ccm-manager address 177.1.10.10 ccm-manager credentials hidden kqp8kECeSyAj1Zqu00cTvQ4E0vzCD5YHSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9 J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmP end subsystem ccn trigger jtapi phonenumber 4220 application voicemail enabled maxsessions 4 end trigger voicemail mailbox owner SCPH1 size 5538 end mailbox This is output from show ccn subsystem jtapi Cisco Call Manager: 177.1.10.10 CCM JTAPI Username: cuejtapi CCM JTAPI Password: * Call Control Group 1 CTI ports: 4221,4222,4223 Call Control Group 1 MWI port: CSS for redirects from route points:ccm-default CSS for redirects from CTI ports: redirecting-party On the CUCM i created three CTI route point named them 4221, 4222 and 4223 and added DN's with the same number to them. I also created an application user named cuejtapi gave it a password and added it to the CTI enabled group . I also created a new voice pilot for 4420. I then created a voice mail profile and added the pilot to it i then assigned the profile to the phone I can ping the CUCM from the CUE Any ideas what i am doing wrong and which commands would you us to troubleshoot. With kind regards, joli-coeur Wouter ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Em on CME
You add that URL under telephony-service. After that you'll have to reset the phones to download the new config with that URL. Jeff -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of iptuse...@hotmail.co.uk Sent: Wednesday, January 12, 2011 1:01 PM To: ccie voice Subject: Re: [OSL | CCIE_Voice] Em on CME Hi I looked through the admin and is just states authenticate url http. How does this get assigned to the services key as with cucm. In cucm 6 and lower you assign the service to the key/user how does the user access em Tamer Ismail tih...@gmail.com wrote: What do you mean by service key? You don't have service url? -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of iptuse...@hotmail.co.uk Sent: Wednesday, January 12, 2011 6:51 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Em on CME Im trying test EM on the cme but have an issue with service key not showing through EM service. Can anyone have a step by step guide on through url and getting it assigned to the service key Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CallManager 6.x/7.x web page issue
Can you ping your PC from the server? Can you ping the server from your PC? Make sure you have connectivity first. Then look into other possible issues. Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Chevy Sent: Tuesday, January 11, 2011 5:34 AM To: Rashid Khan Cc: ccie voice Subject: Re: [OSL | CCIE_Voice] CallManager 6.x/7.x web page issue From the command line do a utils service list and make sure the apache service is running. On Jan 11, 2011 6:14 AM, Rashid Khan me_rashid...@yahoo.com wrote: Hi Friend, I have just installed CallManager 6.x/7.x on vmWare, I am able to ping this machine, but unable to access it's webpage.by enter this command https://call manger ip address/ccmadmin I also tried http://call manger ip address/ccmadmin but the error i get from multiple browsers that unable to open that webpage... Kind Regards Rashid. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Inter- Presence CCM-CME
Hi Francesc, Here is what I have in my notes. I know in the past I had issues getting it to work with a CUCM monitoring a CME phone, but the CME monitoring CUCM worked fine. I had written notes that Vik said CUCM SCCP Phone can't monitor CME phones in 7.0, but SIP should work fine. That was most likely my issue. voice service voip sip bind all ip of cme presence presence enable server ip of cucm allow subscribe Configure a SIP trunk on CUCM with the following: - IP address = CME address - Subscribe CSS that sees phones - You may have to create Security Profile with Accept Presence Subscription and/or Accept Unsolicited Notification, but try without it first Configure a Route Pattern that is the extension(s) for CME phones (ie 3XXX) - Gateway/Route List = SIP trunk configured above Next you would have to configure the monitoring on CUCM and/or CME. Let me know if this works for you or not as I can't configure and test right now. Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roig Borrell, Francesc Xavier Sent: Tuesday, January 11, 2011 7:11 AM To: CCIE_Voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Inter- Presence CCM-CME Hi All, I am trying to configure blfs between CME and CCM. I have not been able to find good documentation for this integration. CME offers a way to configure subsribe prsence to external phones using the allow subscribe all command and a server ip address under presence. The configuration for blfs between 2 CMEs works me fine BR2 presence presence call-list server 10.10.210.11 watcher all allow subscribe Does anyone tried it? Any documentation? Any help would be very much aprreciated. Thanks in advance!! Francesc ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] auto provision none versus all
auto provision all will gather ephone and ephone-dn configuration using SNAP and store in the running config. auto provision dn will only gather and store the ephone-dn in running config. auto provision none will not store anything in the running config. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Shrini Sent: Tuesday, January 11, 2011 6:05 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] auto provision none versus all Hi All, I have a question on SRST which command should I use none or all , help is little confusing. What does include both learned means and include NONE of the learned DNs. BR1-RTR(config-telephony)#srst mode auto-provision ? all SRST mode ON (include both learned DNs and phones into show running) dnSRST mode ON (include only learned DNs into show running) none SRST mode ON (include NONE of the learned DNs/ephones into show running) T I A Shrini ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] BUSY Trigger
There is a setting on the directory number configuration page busy-trigger-per-button and max-calls-per-button. The busy trigger will accomplish what you want if you set to 1. It's easier to think of these two settings as channels. Imagine 8 channels and there are 2 active calls on the line. The busy trigger is set to 2 and the max-calls is set to 3. An incoming call will be redirected by the busy settings on the line. If the user places both calls on hold, the user can still make one additional outgoing call. The busy trigger will only apply to incoming calls, but if the channels (no matter if they were incoming calls or outgoing calls) are occupied the busy settings are used. Also, remember that the max-calls applies to the whole phone. Hope this helps, Jeff -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kiyam Kadir Sent: Monday, January 10, 2011 9:11 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] BUSY Trigger Dear Experts, I hope you all are doing fine. I wonder if you can spare a minute and help me troubleshoot a real-life scenario that is giving me a lot of headache. I have a Cisco 3745 running MGCP I have ISDN PRI E1 connections setup with a telco. Isdn is fine, and multi frame is established. Gateway is registered to CCM. When I place a call to an extension on the telco side I see normal call flow. When I place a second call to the same extension that is talking I am expecting from the call manager to detect that the line is busy and Give me a busy trigger which I am about to use in an IPCC script. Unfortunately the second call is still going to alering state and the ccm does not detect the busy PI. I have tested call waiting on the telco side by calling from two extension to the same above and it works just fine and tells me that it is busy, but I cant get ccm to recognize the busy PI. Please advise if you have time. Thank you Best Regards, On 1/10/11 5:46 PM, ccie_voice-requ...@onlinestudylist.com 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. Direct call park -retrieve (Mritunjay Kumar) 2. Re: + display on phone (Matthew Berry) -- Message: 1 Date: Mon, 10 Jan 2011 20:11:38 +0530 From: Mritunjay Kumar mjs...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Direct call park -retrieve Message-ID: aanlktikjyn8oyios3_jzacc45zt+ffxaye60p=71r...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi All, i have direct call park config ephone-dn 1 number park-slot direct i am able to park the call by transferring the call at but when i tried to retried , it gives busy tone. tried using FAC and gPickup soft but did not help :( any missing config Regards, Mritunay -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20110110/1d824bba/attachment-0001.html -- Message: 2 Date: Mon, 10 Jan 2011 08:46:39 -0600 From: Matthew Berry matthew.be...@cdw.com To: Shrini linuxbos...@gmail.com, 'Friderich Claude' cfrider...@netcore.lu, 'Roig Borrell, Francesc Xavier' francesc.ro...@tecnocom.es, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] + display on phone Message-ID: c9507716.9f50%matthew.be...@cdw.com Content-Type: text/plain; charset=windows-1252 Guys - What you're seeing is a known bug with IOS. I believe it's fixed in 15.1(1)T1. During my study process, it was confirmed that you only need to make sure that the + formatted number shows at the bottom of the screen. You can't lose points for a bug that affects the LCD display in one area, but not another. Of course, these is simply my opinion. I don't have a ruling from Ben Ng. Matthew Berry Sr. Voice Engineer - CCIE 26721 [cid:5C344201-1E3E-4C99-9ABB-886543B761D0]http://www.cdw.com/content/serv ices/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: Shrini linuxbos...@gmail.commailto:linuxbos...@gmail.com Date: Mon, 10 Jan 2011 05:03:26 -0600 To: 'Friderich Claude' cfrider...@netcore.lumailto:cfrider...@netcore.lu, 'Roig Borrell, Francesc Xavier'
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 56, Issue 198
You have to create an Alternative Number for that user in CUC. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of akash patel Sent: Friday, January 07, 2011 2:58 PM To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 56, Issue 198 As it relates For and By for SRST/CUFR, I tried changing VM mask to full E164, but then VM would not work (would get generic UC greeting). Did someone find out a way around? Thank you _ From: ccie_voice-requ...@onlinestudylist.com ccie_voice-requ...@onlinestudylist.com To: ccie_voice@onlinestudylist.com Sent: Sat, October 30, 2010 6:20:54 PM Subject: CCIE_Voice Digest, Vol 56, Issue 198 Note: Forwarded message is attached. 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: Vol2 Lab8 CUBE Xcoder (Brian Mulgrew) 2. Query on Soft-Pones Features (Mann Chaddha) 3. Re: Call Forward Unregistered (=?gbk?B?YnJ1bm8=?=) 4. Re: Vol2 Lab8 CUBE Xcoder (Amr Sherif) ___ CCIE_Voice mailing list CCIE_Voice@onlinestudylist.com http://onlinestudylist.com/mailman/listinfo/ccie_voice ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] T1 Pri Issue
Do you have this GW configured on CUCM? Is the gateway showing registered on CUCM? Do you have mgcp configured/enabled on the router? Have you bounced the MGCP (no mgcp/mgcp) after configuring the pri? You will see TEI Assigned until you have successfully configured all aspects of the MGCP GW. Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Deepak sidana Sent: Thursday, January 06, 2011 1:25 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] T1 Pri Issue Hi All, I am trying to connect the T1 from Br1-RTR to PSTN-WAN. Only when i use service mgcp, under controller, layer 2 isdn staus as TEI_ASSIGNED. At PSTN-WAN Router, i am using isdn protocol-emulate network under s0/0/0:23 Branch1 Config:- BR1-RTR#sh isdn sta Global ISDN Switchtype = primary-net5 ISDN Serial0/0/0:23 interface dsl 0, interface ISDN Switchtype = primary-net5 Layer 1 Status: ACTIVE Layer 2 Status: TEI = 0, Ces = 1, SAPI = 0, State = TEI_ASSIGNED ! controller T1 0/0/1 framing esf linecode b8zs cablelength long 0db ! interface Serial0/0/0:23 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn bind-l3 ccm-manager isdn incoming-voice voice no cdp enable Please share you experince, if some one faced the same issue. ThanksRgds Deepak Sidana. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Mobile Voice Access
Hi Shrini, If you follow the Features and Services Guide as mentioned before, you will have success. You need to configure hairpinning for MGCP to work with MVA. The idea is that you will accept the call using 5999, but the MVA pilot number will be a different number. You will have to add the h323 gateway to CUCM and create a route pattern that will direct calls for 5999 to the h323 gateway. The incoming dialpeer on the gateway will be the 5999 number which will trigger the VXML script. There will be a different number needed for the MVA pilot, for example 6000. The outgoing dial-peer will point back to CUCM using this number (6000). As the h323 gateway and the MGCP gateway will be logically separate (listening to different interfaces), you can accomplish this on the same box. Hope this helps, Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roger Carpio Sent: Sunday, January 02, 2011 7:40 AM To: ShinGei Yong Cc: ccie_voice@onlinestudylist.com; Shrini Subject: Re: [OSL | CCIE_Voice] Mobile Voice Access Hello ShinGei, Thanks for the info. In the end, MGCP will not make it for us Regards, Roger Carpio. On Sun, Jan 2, 2011 at 9:34 AM, ShinGei Yong shingei.y...@gmail.com wrote: Hi Roger, MVA on a MGCP control gateway is possible.In fact,that is a coexisting of both MGCP and H323 on the same gateway,but you could not used a MGCP control PRI for MVA. you may refer to Netpro for own interest. https://supportforums.cisco.com/thread/2005673 Shingei. On Sun, Jan 2, 2011 at 11:16 PM, Shrini linuxbos...@gmail.com wrote: Hi ShinGei , bkvalentine, Rogers et al I remember it was successful last time when I configured it another lab when HQ was h323. Now I was confused around dial-peers hence had the question. I will give a try now with MGCP + H323 on HQ and it should work. Thanks all. Shrini From: ShinGei Yong [mailto:shingei.y...@gmail.com] Sent: Sunday, January 02, 2011 6:42 AM To: Shrini; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Mobile Voice Access Hi Shrini, What about your UCM configuration? 1. is your H323 GW registered with UCM? 2. what is your dialing behavior internally?4 or 10?if is 4, then your in outbound dp should be 4 digit patten as well instead of 10. Please provide more info Shingei On Sun, Jan 2, 2011 at 4:13 PM, Shrini linuxbos...@gmail.com wrote: Hi Experts, Wish you all a Happy and Prosperous New Year 2011 First question this year :-) HQ Site is MGCP. When I call HQ Phone 5002 , HQ PSTN is ringing -- all is good. I have configured MVA number 5999 in service parameters and Media Resources -- MVA -- 5999 / PT-INTERNAL / English on router. application service cmm http://177.1.10.10:8080/ccmivr/pages/IVRMainpage.vxml ! ! dial-peer voice 5999 pots service cmm incoming called-number 2123945999 no digit-strip Also on CUCM : But when I call 2123945999 from HQ PSTN 2123942123 I am getting fast busy tone. It is not invoking the vxml script. What am I doing wrong here ? TIA Shrini ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com image001.gif___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation
Hi Francesc, A payload of 20 and 10 is not correct. RSVP and LLQ calculations are two different things. For RSVP, you calculations are correct. Correct Payloads (20 ms) G711 - 160 G729 - 20 For example, FRF.12, G729, with compression: IP/UDP/RTP - 2 bytes G729 - 20 bytes FRF.12 - 8 bytes 2 + 20 + 8 = 30 bytes per packet 30 bytes * 8 bits = 240 bits per packet 240 bits per packet * 50 packets per second = 12000 bits per second or 12 Kbps FRF.12, G729 without compression: IP/UDP/RTP = 40 bytes G729 - 20 bytes FRF.12 - 8 bytes 40 + 20 + 8 = 68 bytes per packet 68 * 8 = 544 bits per packets 544 bpp * 50 packets per second = 27200 bits per second or 27.2 Kbps FRF.12, G711 without compression: IP/UDP/RTP = 40 G711 = 160 FRF.12 - 8 40 + 160 + 8 = 208 bytes per packets 208 * 8 = 1664 bpp 1664 * 50 pps = 83200 bps or 83.2 Kbps Hope this helps, Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roig Borrell, Francesc Xavier Sent: Wednesday, January 05, 2011 7:42 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] RSVP LLQ priority value calculation Hi everyone! I am trying to understand the right way to calculate the priority value in LLQ with a RSVP configuration. I have not been able to find documentation clarifying this. So supposing HQ-BR1 4 calls g729 ip rsvp bandwitdh = 24*3 + 40 = 112 No problem with the rsvp bandwith, 3 calls with 20ms sample rate and one call with the worst case 10ms sample rate. So following this and considering FR12 . The priority queue should be calculated this way L2 7 L2 7 L3 40 L3 40 Payload 20 Payload 10 67*8*50= 26,8kbps57*8*100 = 45,6kbps LLQ priority = 28,6*3 + 45,6 = 131,4 -132 Do you agree? Is it the right way? Thanks in advance! Francesc ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation
Hi Francesc, As I noted before, the RSVP bandwidth calculation is different from the LLQ bandwidth calculation. For the scenario of 2 RSVP calls, you will need to calculate as follows: 40 + 24 = 64 (one worst case 10ms call and one normal 20 ms) So under the serial interfaces you will configure ip rsvp bandwidth 64 The question states that you need to put the RSVP traffic in the PQ. This means that the traffic will have to be marked as EF to make it into the LLQ. Under the same serial interface, enter the ip rsvp signaling ef command Now you need to calculate your BW for the LLQ. IP/UDP/RTP - 40 Payload 20 FRF.12 8 40 + 20 + 8 = 68 68 bytes * 8 bits = 544 bits per packet 544 bpp * 50 pps = 272000 bps or 27.2 Kbps 2 G729 calls * 27.2 Kbps = 54.4 Kbps or roughly 55 Kbps A basic LLQ without RSVP overhead would need to have a priority 55 command. However, the question asks for you to take this extra overhead for RSVP into account. IP/UDP/RTP - 40 Payload 10 FRF.12 8 40 + 10 + 8 = 58 bytes 58 * 8 = 464 bpp 464 * 100 pps = 46400 bps or 46.4 Kbps Therefore the bandwidth calculation would instead be 27.2 + 46.4 = 73.6 Kbps or 74 Kbps. Hope this helps, Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roig Borrell, Francesc Xavier Sent: Wednesday, January 05, 2011 10:10 AM To: Shrini Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation Hi Shrini, Thank you for your answer. I dont see very clear how you take into consideration L2 header These values 40kbps (g729 10ms)/ 24kbps (g729 20ms) only consider L3+UDP/RTP+Payload. For 2 g729 calls ip rsvp bandwith= 24+40 =64 OK But which value would you use for priority queue if you have this question Between HQ-BR1 provision enough bandwidth in the priority queue for 2 calls. Any RSVP traffic should be placed into the PQ. Ensure that you provision additional amount of bandwidth in the PQ to include RSVP traffic Thanks!! Francesc Lets take example of one g729 call rsvp a= Worst case = (L3+PL)*8*100 = (40+10)*8*100 = 40k b= One G729 = ((L2+L3)+PL)*50*8 = ((40)+20)*8*50 = 24 L2+L3 = IP+RTP+UDP = 40 De: Shrini [mailto:linuxbos...@gmail.com] Enviado el: miércoles, 05 de enero de 2011 18:33 Para: Roig Borrell, Francesc Xavier CC: ccie_voice@onlinestudylist.com Asunto: Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation Hi Roig, Each call in Location based CAC : g729 - 24k g711 - 80k RSVP: g729 - 40k g711 - 96k Gatekeeper: g729 - 16k g711 - 128k (not sure 100%) For your question Lets take example of one g729 call rsvp a= Worst case = (L3+PL)*8*100 = (40+10)*8*100 = 40k b= One G729 = ((L2+L3)+PL)*50*8 = ((40)+20)*8*50 = 24 L2+L3 = IP+RTP+UDP = 40 So one call need 64k but 40 is hardcoded for rsvp. Thanks Shrini On 1/5/2011 7:42 AM, Roig Borrell, Francesc Xavier wrote: Hi everyone! I am trying to understand the right way to calculate the priority value in LLQ with a RSVP configuration. I have not been able to find documentation clarifying this. So supposing HQ-BR1 4 calls g729 ip rsvp bandwitdh = 24*3 + 40 = 112 No problem with the rsvp bandwith, 3 calls with 20ms sample rate and one call with the worst case 10ms sample rate. So following this and considering FR12 . The priority queue should be calculated this way L2 7 L2 7 L3 40 L3 40 Payload 20 Payload 10 67*8*50= 26,8kbps57*8*100 = 45,6kbps LLQ priority = 28,6*3 + 45,6 = 131,4 -132 Do you agree? Is it the right way? Thanks in advance! Francesc ___ 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] RSVP LLQ priority value calculation
I looked at the PG and they add in the calculation as I detailed in my most recent email. However, I am totally with you. The RTP/LLQ is different from the RSVP CAC and I would think that only a few extra Kbps would account for the RSVP control traffic in the PQ. Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Miron Kobelski Sent: Wednesday, January 05, 2011 10:49 AM To: Roig Borrell, Francesc Xavier Cc: ccie_voice@onlinestudylist.com; Shrini Subject: Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation Hi, RSVP reservation and actual LLQ usage are 2 different things. I think you should keep in mind, that there is no traffic in PQ before RSVP reservation completes. For RSVP calculation you only take into account L3. You have 2 possible bandwidth values: * standard (24kbps for G729/20ms) and * worst case (40 kbpbs for G729/10ms), because when the destination is ringing capabilities exchange has not yet occured and there is no media flow. That's why at this stage worst case is assumed (g729/40ms). PQ is still empty. As soon as the call is answered, capabilities are exchanged and decision about codec/payload is made - reservation can be decreased to standard 24kbps (g729/20ms). Only now the RTP flow can occur - PQ is filled up and served by LLQ (with values calculated including L2 overhead). One more thing - the task requirement is not very clear: RSVP traffic for me consists only of those several small RSVP protocol messages exchanged during RSVP negotiation. I'd not include RTP traffic in it... So I guess 5kbps should be more than enough. Anybody disagrees? HTH kobel On Wed, Jan 5, 2011 at 19:10, Roig Borrell, Francesc Xavier francesc.ro...@tecnocom.es wrote: Hi Shrini, Thank you for your answer. I don’t see very clear how you take into consideration L2 header These values 40kbps (g729 10ms)/ 24kbps (g729 20ms) only consider L3+UDP/RTP+Payload. For 2 g729 calls ip rsvp bandwith= 24+40 =64 OK But which value would you use for priority queue if you have this question Between HQ-BR1 provision enough bandwidth in the priority queue for 2 calls. Any RSVP traffic should be placed into the PQ. Ensure that you provision additional amount of bandwidth in the PQ to include RSVP traffic Thanks!! Francesc ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation
Definitely, Im sorry I didnt understand at first J Happy studies! Jeff From: Roig Borrell, Francesc Xavier [mailto:francesc.ro...@tecnocom.es] Sent: Wednesday, January 05, 2011 12:12 PM To: givemeccievoice2...@gmail.com; 'Shrini' Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] RSVP LLQ priority value calculation Hi Jeff, Great! Then we agree with the solution for this requirement. J Thank you very much!! De: givemeccievoice2...@gmail.com [mailto:givemeccievoice2...@gmail.com] Enviado el: miércoles, 05 de enero de 2011 20:53 Para: Roig Borrell, Francesc Xavier; 'Shrini' CC: ccie_voice@onlinestudylist.com Asunto: RE: [OSL | CCIE_Voice] RSVP LLQ priority value calculation Hi Francesc, As I noted before, the RSVP bandwidth calculation is different from the LLQ bandwidth calculation. For the scenario of 2 RSVP calls, you will need to calculate as follows: 40 + 24 = 64 (one worst case 10ms call and one normal 20 ms) So under the serial interfaces you will configure ip rsvp bandwidth 64 The question states that you need to put the RSVP traffic in the PQ. This means that the traffic will have to be marked as EF to make it into the LLQ. Under the same serial interface, enter the ip rsvp signaling ef command Now you need to calculate your BW for the LLQ. IP/UDP/RTP - 40 Payload 20 FRF.12 8 40 + 20 + 8 = 68 68 bytes * 8 bits = 544 bits per packet 544 bpp * 50 pps = 272000 bps or 27.2 Kbps 2 G729 calls * 27.2 Kbps = 54.4 Kbps or roughly 55 Kbps A basic LLQ without RSVP overhead would need to have a priority 55 command. However, the question asks for you to take this extra overhead for RSVP into account. IP/UDP/RTP - 40 Payload 10 FRF.12 8 40 + 10 + 8 = 58 bytes 58 * 8 = 464 bpp 464 * 100 pps = 46400 bps or 46.4 Kbps Therefore the bandwidth calculation would instead be 27.2 + 46.4 = 73.6 Kbps or 74 Kbps. Hope this helps, Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roig Borrell, Francesc Xavier Sent: Wednesday, January 05, 2011 10:10 AM To: Shrini Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation Hi Shrini, Thank you for your answer. I dont see very clear how you take into consideration L2 header These values 40kbps (g729 10ms)/ 24kbps (g729 20ms) only consider L3+UDP/RTP+Payload. For 2 g729 calls ip rsvp bandwith= 24+40 =64 OK But which value would you use for priority queue if you have this question Between HQ-BR1 provision enough bandwidth in the priority queue for 2 calls. Any RSVP traffic should be placed into the PQ. Ensure that you provision additional amount of bandwidth in the PQ to include RSVP traffic Thanks!! Francesc Lets take example of one g729 call rsvp a= Worst case = (L3+PL)*8*100 = (40+10)*8*100 = 40k b= One G729 = ((L2+L3)+PL)*50*8 = ((40)+20)*8*50 = 24 L2+L3 = IP+RTP+UDP = 40 De: Shrini [mailto:linuxbos...@gmail.com] Enviado el: miércoles, 05 de enero de 2011 18:33 Para: Roig Borrell, Francesc Xavier CC: ccie_voice@onlinestudylist.com Asunto: Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation Hi Roig, Each call in Location based CAC : g729 - 24k g711 - 80k RSVP: g729 - 40k g711 - 96k Gatekeeper: g729 - 16k g711 - 128k (not sure 100%) For your question Lets take example of one g729 call rsvp a= Worst case = (L3+PL)*8*100 = (40+10)*8*100 = 40k b= One G729 = ((L2+L3)+PL)*50*8 = ((40)+20)*8*50 = 24 L2+L3 = IP+RTP+UDP = 40 So one call need 64k but 40 is hardcoded for rsvp. Thanks Shrini On 1/5/2011 7:42 AM, Roig Borrell, Francesc Xavier wrote: Hi everyone! I am trying to understand the right way to calculate the priority value in LLQ with a RSVP configuration. I have not been able to find documentation clarifying this. So supposing HQ-BR1 4 calls g729 ip rsvp bandwitdh = 24*3 + 40 = 112 No problem with the rsvp bandwith, 3 calls with 20ms sample rate and one call with the worst case 10ms sample rate. So following this and considering FR12 . The priority queue should be calculated this way L2 7 L2 7 L3 40 L3 40 Payload 20 Payload 10 67*8*50= 26,8kbps57*8*100 = 45,6kbps LLQ priority = 28,6*3 + 45,6 = 131,4 -132 Do you agree? Is it the right way? Thanks in advance! Francesc ___ 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] RSVP LLQ priority value calculation
After I just agreed with you! J Below is not the RSVP calculation. That is the LLQ bandwidth calculations. After I reviewed my notes and figured out the value necessary, I referred to the PG. The PG calculates the PQ bandwidth by using 1 call at 10ms and 1 call at 20ms. I am confused as to why they do it this way. I would think that you would use the 27.2 Kbps for each call and arrive at a 55 Kbps BW in the LLQ. I agree with you that the RSVP communications will only require minimal overhead and you can just simply add a couple of Kbps to accomplish this task. Remember, the question that Francesc was referring to assumes you have RSVP configured already, and is asking you to configure the LLQ including the necessary overhead for RSVP messages. Jeff From: Miron Kobelski [mailto:findko...@gmail.com] Sent: Wednesday, January 05, 2011 1:13 PM To: givemeccievoice2...@gmail.com Cc: Roig Borrell, Francesc Xavier; Shrini; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation I disagree... I would never include L2 in RSVP bandwidth calculations. To see what values RSVP uses, check show ip rsvp installed in ringing and connected states. it is 40 and 24 kbps for g729. I'd say that RSVP overhead should constitute no more then 1kbps (only several small messages during RSVP negotations!) regards kobel On Wed, Jan 5, 2011 at 20:53, givemeccievoice2...@gmail.com wrote: FRF.12 – 8 40 + 20 + 8 = 68 68 bytes * 8 bits = 544 bits per packet 544 bpp * 50 pps = 272000 bps or 27.2 Kbps 2 G729 calls * 27.2 Kbps = 54.4 Kbps or roughly 55 Kbps A basic LLQ without RSVP overhead would need to have a priority 55 command. However, the question asks for you to take this extra overhead for RSVP into account. IP/UDP/RTP - 40 Payload – 10 FRF.12 – 8 40 + 10 + 8 = 58 bytes 58 * 8 = 464 bpp 464 * 100 pps = 46400 bps or 46.4 Kbps Therefore the bandwidth calculation would instead be 27.2 + 46.4 = 73.6 Kbps or 74 Kbps. Hope this helps, Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation
Hi Shrini, I believe you’re correct as well, but you were detailing the RSVP BW calculation not the LLQ which the question was asking. Jeff From: Shrini [mailto:linuxbos...@gmail.com] Sent: Wednesday, January 05, 2011 3:32 PM To: Roig Borrell, Francesc Xavier Cc: givemeccievoice2...@gmail.com; 'Miron Kobelski'; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation Thanks for the details debugs Jeff. Just wanted to double check with you that my examples are also correct ? Thanks again Shrini On 1/5/2011 2:42 PM, Roig Borrell, Francesc Xavier wrote: Hi guys, Yes, thinking twice it doesn’t make a lot of sense consider the call with the worst case payload (46.4) in order to adding RSVP signaling. 1 RSVP Request Dec 17 18:47:58.630: RSVP 10.10.110.2_16548-10.10.110.1_17938[0.0.0.0]: start requesting 40 kbps FF reservation for 10.10.110.2 2 RSVP update (Call established ) Dec 17 18:49:10.047: RSVP-RESV: Locally created reservation. No admission/traffic control needed Dec 17 18:49:10.047: RSVP 10.10.110.2_16510-10.10.110.1_19416[0.0.0.0]: start requesting 24 kbps FF reservation for 10.10.110.2 In fact in the first step, there isn’t RTP traffic, so in case of congestion the PQ only will have some RSVP packets. So the requirement can be achieved PQ = 27,2*2 + X, X (extra RSVP signaling traffic, as Miron we can consider 1kbps) Now, I believe we all agree!! J Thanks for your help! Happy studies! Francesc De: givemeccievoice2...@gmail.com [mailto:givemeccievoice2...@gmail.com] Enviado el: miércoles, 05 de enero de 2011 22:42 Para: 'Miron Kobelski' CC: Roig Borrell, Francesc Xavier; 'Shrini'; ccie_voice@onlinestudylist.com Asunto: RE: [OSL | CCIE_Voice] RSVP LLQ priority value calculation After I just agreed with you! J Below is not the RSVP calculation. That is the LLQ bandwidth calculations. After I reviewed my notes and figured out the value necessary, I referred to the PG. The PG calculates the PQ bandwidth by using 1 call at 10ms and 1 call at 20ms. I am confused as to why they do it this way. I would think that you would use the 27.2 Kbps for each call and arrive at a 55 Kbps BW in the LLQ. I agree with you that the RSVP communications will only require minimal overhead and you can just simply add a couple of Kbps to accomplish this task. Remember, the question that Francesc was referring to assumes you have RSVP configured already, and is asking you to configure the LLQ including the necessary overhead for RSVP messages. Jeff From: Miron Kobelski [mailto:findko...@gmail.com] Sent: Wednesday, January 05, 2011 1:13 PM To: givemeccievoice2...@gmail.com Cc: Roig Borrell, Francesc Xavier; Shrini; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation I disagree... I would never include L2 in RSVP bandwidth calculations. To see what values RSVP uses, check show ip rsvp installed in ringing and connected states. it is 40 and 24 kbps for g729. I'd say that RSVP overhead should constitute no more then 1kbps (only several small messages during RSVP negotations!) regards kobel On Wed, Jan 5, 2011 at 20:53, givemeccievoice2...@gmail.com wrote: FRF.12 – 8 40 + 20 + 8 = 68 68 bytes * 8 bits = 544 bits per packet 544 bpp * 50 pps = 272000 bps or 27.2 Kbps 2 G729 calls * 27.2 Kbps = 54.4 Kbps or roughly 55 Kbps A basic LLQ without RSVP overhead would need to have a priority 55 command. However, the question asks for you to take this extra overhead for RSVP into account. IP/UDP/RTP - 40 Payload – 10 FRF.12 – 8 40 + 10 + 8 = 58 bytes 58 * 8 = 464 bpp 464 * 100 pps = 46400 bps or 46.4 Kbps Therefore the bandwidth calculation would instead be 27.2 + 46.4 = 73.6 Kbps or 74 Kbps. Hope this helps, Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RSVP CAC
Hi Shrini, CUCM will always request the worst case scenario in bandwidth first. The easy way to do this is to increase the max bandwidth command and use the show ip rsvp bandwidth command. For example: 1. Increase the ip rsvp bandwidth command to 120 2. Dial a SB phone from HQ, but don't answer 3. Run the show ip rsvp bandwidth command. This will show you the worst case that is being requested by CUCM (in this case 40) 4. Answer the call 5. Run the show ip rsvp bandwidth command, you will see it has dropped back to 24. 6. You then can calculate the value as for 2 calls as 40 +24, or 3 calls 40 + 24 + 24, etc. For your scenario, the value should be ip rsvp bandwidth 64. If this still isn't working then there is something else wrong. Consider these: - If you use MLPP for WAN QoS, you need to move the rsvp command under the Virtual interface. - The ip rsvp bandwidth command should be on both WAN interfaces - Restart the Devices after changing the Location values (although this shouldn't matter, it's still worth a shot) - Run debug ip rsvp and see that RSVP is even being used (although if you can't see output in the show ip rsvp bandwidth as suggested above that will show it as well) - Ensure that the MTPs are registered and in the correct MRGs and MRGLs, reset the MTPs, reset devices in the DPs, etc For help with the debug statements, Matt did a good job of detailing them here and provided a working router configuration - http://matthewberry.info/ciscovoiceguru/377/debug-ip-rsvp-messages/ Hope this helps and Merry Christmas, Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Prashant Patel Sent: Friday, December 24, 2010 11:54 AM To: Shrini Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] RSVP CAC Hi Shrini, Do you have the rsvp command in the mtp configuration? HTH Prashant On Fri, Dec 24, 2010 at 2:06 PM, Shrini linuxbos...@gmail.com wrote: Hi Experts, Wish you all and your families MERRY CHRISTMAS and HAPPY HOLIDAYS Below is the scenario. From HQ to BR1 I want to allow only 2 G729 calls. HQ to BR1 call are G729 What I did is : Created MTP on HQ and BR1 routers. dsp profile 2 mtp codec g729r8 codec pass-through maximum sessions soft 2 associate application SCCP and registered them with CUCM. Under serial interfaces: ip rsvp bandwidth 80 On Call manager: HQ - BR1 - Audio ULTD - RSVP setting Mandatory ..vice versa Assigned these locations to DP_HQ and DP_BR1. When I make a call from HQ to BR1 the first call itself says not enough bandwidth My understanding is: Each g729 call need 40K , ip rsvp b/w 80 means 2 calls max session soft 2 - again 2 calls maximum. Why am I getting not enough bandwidth ? T I A Shrini ___ 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
Re: [OSL | CCIE_Voice] Volume 2 Lab 5 Task 2.3
Hi Brian, I would change back their pattern as they are testing you on the following concepts. When the call comes into the GW on CUCM, the prefix values configured will be added to the front of the number. In order to figure out how you will prefix this you need to look at the debug isdn q931 and work with what you've got. Meaning, if you are receiving a Subscriber number 2059432785 then you need to manipulate the Subscriber prefix. This will effect what you will see in the Missed Calls and solve part 2. Now, the phone's device pool has a calling party transformation calling search space applied to it. This will be used incoming on the phone and determine how you will localize the calls on the screen. You will need to have a Calling Party Xformation pattern that will manipulate it back down. This effects the screen and nothing else. I'm trying not to give the answer as understanding these concepts is key and you can solve this in 2 mins once you understand how these all work together. Try and fool around with these values and let us know how it goes. Hope this helps, Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Brian Rudy Sent: Monday, December 20, 2010 8:04 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Volume 2 Lab 5 Task 2.3 I have concluded the preceding 0 that comes in from the PSTN was incorrect in the WB and PG? These are the tasks where it talks about having +44 020 5943 2785 in the missed/received calls directory. First, I was not getting the 0 or National in my debug isdn q931 on inbound calls from the 2nd Button on PSTN Phone to HQ Phone 2. I changed the translation pattern on the PSTN router FROM rule 2 /^2059432785$/ /\0/ type any subscriber plan any isdn TO rule 2 /^2059432785$/ /0\0/ type any national plan any isdn However, if i strip the 0 (National Number - 44:1) at the HQ Gateway and Globalize the Number, then how is going to show in the missed/received calls directory as +44 020 5943 2785? Any insight to this is greatly appreciated! Brian ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CFUR display For and By
I know this doesn't make much sense, but it is done through a VM Profile. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of study2b ccie Sent: Thursday, December 09, 2010 10:10 AM To: OSL Subject: [OSL | CCIE_Voice] CFUR display For and By Hello experts, I am working on CFUR and have a question: When BR1 phones go into SRST mode(I am using call-manager-fallback), I can get HQ phones ring to BR1 Phone 2 On the BR1 Phone 2 screen, it showed: From 2123945002 (by 1002) Instead of show by 1002, what do I need to do for it to show a globalized number? From 2123945002 (by +16178631002) Thank you, ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] H323-CUCM strange behavior
Actually, I don't see a gateway command. Try adding that next time and calls should work with CUCM. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of study2b ccie Sent: Tuesday, December 07, 2010 6:18 PM To: OSL Subject: [OSL | CCIE_Voice] H323-CUCM strange behavior Hello Experts, I was on the vRack today and experienced a strange behavior between H323 and CUCM. BR1 router was running H323, connected to CUCM. All calls, both inbound and outbound between BR1 and PSTN, failed, they rang once and then get dropped. Failed message under Debug ISDN Q931 was: Bearer capability not implemented When I put the phones into SRST mode, all inbound and outbound calls worked fine. I could see ISDN status is up with multiple frames established. The router was behaving normally in most ways. Below is my BR1 configuration, can you help to take a look and let me know what I had missed? Thank you in advance. sh run Building configuration... hostname BR1-RTR ! clock timezone EST -5 clock summer-time EDT recurring network-clock-participate wic 0 network-clock-select 1 T1 0/0/0 ! ! no ip domain lookup ip domain name proctorlabs.com no ipv6 cef ! multilink bundle-name authenticated ! ! isdn switch-type primary-ni ! ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! voice translation-rule 1 rule 1 /.*\($\)/ /\1/ ! voice translation-rule 999 rule 15 // /\0/ type any subscriber plan any isdn ! voice translation-rule 1000 rule 10 // /\0/ type national national plan isdn isdn rule 15 // /\0/ type any subscriber plan any isdn ! voice translation-rule 1001 rule 15 // /+\0/ type any international plan any isdn ! voice translation-rule rule 15 // /\0/ type any unknown plan any isdn ! voice translation-rule 91000 rule 15 // /\0/ type any subscriber plan any isdn ! voice translation-rule 91001 rule 15 // /\0/ type any national plan any isdn ! voice translation-rule 91002 rule 15 // /\0/ type any international plan any isdn ! ! voice translation-profile 1 translate called 1 ! voice translation-profile 1000 translate calling 1000 translate called 91000 ! voice translation-profile 1001 translate calling 1001 translate called 91001 ! voice translation-profile 1002 translate calling 1002 translate called 91002 ! voice translation-profile 999 translate calling 999 translate called ! ! voice-card 0 no dspfarm ! ! ! ! ! archive log config hidekeys ! ! ! ! ! controller T1 0/0/0 framing esf linecode b8zs pri-group timeslots 1-3,24 ! controller T1 0/0/1 framing esf linecode b8zs channel-group 0 timeslots 1-24 ! ! interface Serial0/0/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bchan-number-order ascending isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable ! interface Vlan240 ip address 10.10.201.1 255.255.255.0 ip helper-address 10.10.210.10 h323-gateway voip interface h323-gateway voip bind srcaddr 10.10.201.1 ! ! ! ! ! control-plane ! ! ! voice-port 0/0/0:23 translation-profile incoming 1 ! ! ! ! ! dial-peer voice 1 pots incoming called-number .T direct-inward-dial port 0/0/0:23 ! dial-peer voice 2 voip destination-pattern 1... session target ipv4:10.10.210.11 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 3 voip preference 1 destination-pattern 1... session target ipv4:10.10.210.10 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 999 pots translation-profile outgoing 999 destination-pattern 9%911$ no digit-strip direct-inward-dial port 0/0/0:23 forward-digits 3 ! dial-peer voice 1000 pots translation-profile outgoing 1000 destination-pattern 9[2-9]..$ no digit-strip direct-inward-dial port 0/0/0:23 forward-digits 7 ! dial-peer voice 1001 pots translation-profile outgoing 1001 destination-pattern 91[2-9].$ no digit-strip direct-inward-dial port 0/0/0:23 forward-digits 11 ! dial-peer voice 1002 pots translation-profile outgoing 1002 destination-pattern 9011T direct-inward-dial port 0/0/0:23 prefix 011 ! ! ! ! gatekeeper shutdown ! ! telephony-service srst mode auto-provision all max-ephones 2 max-dn 20 preference 3 no-reg ip source-address 10.10.110.2 port 2000 max-conferences 8 gain -6 transfer-system full-consult transfer-pattern .T secondary-dialtone 9 create cnf-files version-stamp 7960 Dec 07 2010 10:14:21 ! ! ephone-dn 1 octo-line number 1002 label 1002 description +16178631002 name +16178631002 preference 3 ! ! ephone 1 device-security-mode none mac-address 001D.E041.DE1B type 7961 button 1:1 ! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Service Parameters Configuration in CUCM
Randall, When on the Service Parameters page, either click on the actual parameter link or go to Help This Page That is the best and easiest way to see what a specific parameter is all about. Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Randall Crumm Sent: Friday, December 03, 2010 2:51 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Service Parameters Configuration in CUCM HI, Can someone let me know the document that describes the service parameters in CUCM. For example, what if I couldn't remember what the service parameter SIP station keepalive interval actually is and why you would want to change it. I can't find something that describes each parameter. Thanks, Randall Legal Disclaimer: The information contained in this message may be privileged and confidential. It is intended to be read only by the individual or entity to whom it is addressed or by their designee. If the reader of this message is not the intended recipient, you are on notice that any distribution of this message, in any form, is strictly prohibited. If you have received this message in error, please immediately notify the sender and delete or destroy any copy of this message ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] URL's for phone services
Randall, There is no central location for the phone services you may need for the exam. You’ll have to do throughout the Cisco.com documentation. IPMA can be found in the Help provided in CUCM, just search for IPMA and look at the checklist for the IPMA Proxy Line mode. IPPA can be found in the UCCX documentation under the configuration examples for One Button login or the Installation guide for CAD IPPM is in the SRND, just do a search for it. Etc… You’ll just have to find them in the appropriate documents and remember how you go to them for the exam J Hope this helps, Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Randall Crumm Sent: Wednesday, December 01, 2010 10:32 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] URL's for phone services HI, Where can I find a list of URL’s for phone services? Thanks, Randall Legal Disclaimer: The information contained in this message may be privileged and confidential. It is intended to be read only by the individual or entity to whom it is addressed or by their designee. If the reader of this message is not the intended recipient, you are on notice that any distribution of this message, in any form, is strictly prohibited. If you have received this message in error, please immediately notify the sender and delete or destroy any copy of this message ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] URL's for phone services
I believe you can find that stuff in the Release Notes. I know that's where you find the sql statement to insert the Voicemail button functionality back into CUCM. HTH From: khaled Saholy [mailto:khaled_sah...@hotmail.com] Sent: Wednesday, December 01, 2010 12:53 PM To: randall.cr...@flextronics.com; findko...@gmail.com; givemeccievoice2...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] URL's for phone services Hi, What about the other phone services like Fast Dial and Missed Calls ...ect? Where we can find them? I searched for the missed calls service in the net but no luck yet. Regards. _ Date: Wed, 1 Dec 2010 12:16:39 -0800 From: randall.cr...@flextronics.com To: findko...@gmail.com; givemeccievoice2...@gmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] URL's for phone services Thanks Jeff and Miron. I appreciate the assistance. This is very helpful. Randall From: Miron Kobelski [mailto:findko...@gmail.com] Sent: Wednesday, December 01, 2010 11:55 AM To: Randall Crumm Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] URL's for phone services Hi, http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg18379.html regards kobel On Wed, Dec 1, 2010 at 19:31, Randall Crumm randall.cr...@flextronics.com wrote: HI, Where can I find a list of URL's for phone services? Thanks, Randall Legal Disclaimer: The information contained in this message may be privileged and confidential. It is intended to be read only by the individual or entity to whom it is addressed or by their designee. If the reader of this message is not the intended recipient, you are on notice that any distribution of this message, in any form, is strictly prohibited. If you have received this message in error, please immediately notify the sender and delete or destroy any copy of this message ___ 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] 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.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mritunjay Kumar Sent: Tuesday, November 30, 2010 3:06 AM To: 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: 14.160.110.15:1720 US_1 Zone CUBE master gateway list: 14.160.110.254:1720 MJ-CUBE Prefix: 12#* Zone CUBE master gateway list: 14.160.110.254:1720 MJ-CUBE Prefix: 44#* Zone CME master gateway list: 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.com 14.160.110.129 zone local CME cisco.com invia CUBE outvia CUBE zone local CUCM cisco.com invia CUBE outvia CUBE zone local CUBE cisco.com zone prefix CUCM 2... zone prefix CME 5... gw-type-prefix 1#* no shutdown Is any configmissing ?? Regards, MJ ___ 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
I still think your problem most likely lies in the tech prefix on the CUBE. You don't need a tech prefix and I would make the dial-peer a little more specific. I'm not sure completely sure that this is accurate, but I would think that having a tech prefix on CUBE of 1# would not allow you route a call to it with 44#. I'm not 100% on that though. Also, with the IPIPGW not found error, it normally mean you have to bounce the gateway so that it registers as a IPIPGW. If you type in the gateway command prior to the allow connections command then the GW will not register as an IPIPGW and even if you enter those commands unless it unregisters and registers again, the gatekeeper will not update on its own. Change the config to this: interface GigabitEthernet0/0 ip address 14.160.110.254 255.255.255.0 duplex auto speed auto media-type rj45 h323-gateway voip interface h323-gateway voip id CUBE ipaddr 14.160.110.129 1719 h323-gateway voip h323-id MJ-CUBE h323-gateway voip tech-prefix 1# (remove) h323-gateway voip bind srcaddr 14.160.110.254 voice service voip allow h t h allow s t h allow s t s allow h t s no gateway gateway dial-peer voice 100 voip incoming called-number 44# no vad dtmf-relay h245-signal codec g729r8 dial-peer voice 200 voip destination-pattern 44# session target ras no vad dtmf-relay h245-signal codec g729r8 Try the call again and do a debug gatekeeper main 10 to see if the gateway is selected. If the gateway is selected and the call still fails, then try an debug voip dialpeer on both routers, and also possibly a debug voice ipipgw on HQ. Hope this helps, Jeff From: Mritunjay Kumar [mailto:mjs...@gmail.com] Sent: Tuesday, November 30, 2010 9:28 AM To: Matthew Berry; Prashant Patel; givemeccievoice2...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] call from CUCM to CME : VIA Zone Hi All , Might not be able to check my mails for 1-2 days . I will try to reply in case if any other information is needed. Regards, MJ On Tue, Nov 30, 2010 at 7:51 PM, Mritunjay Kumar mjs...@gmail.com wrote: Hi All, thanks for responding. Here is the scenario , config and result Scenario CME registered with 44# , CUCM and CUBE with 1# Call is faling from CUCM to CME only. output and config Gatekeep config gatekeeper zone local BR1 cisco.com 14.160.110.129 zone local CME cisco.com invia CUBE outvia CUBE zone local CUCM cisco.com invia CUBE outvia CUBE zone local CUBE cisco.com zone prefix CUCM 2... zone prefix CME 5... no shutdown BR1#sh gatek endpoints GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 14.160.110.15 1720 14.160.110.15 32797 CUCM VOIP-GW ENDPOINT-ID: 71A5D3680003 VERSION: 5 AGE: 37 secs SupportsAnnexE: FALSE g_supp_prots: 0x0050 H323-ID: US_1 Voice Capacity Max.= Avail.= Current.= 0 14.160.115.200 1720 14.160.115.200 57052 CME H323-GW ENDPOINT-ID: 70B6E1940003 VERSION: 4 AGE: 13 secs SupportsAnnexE: FALSE g_supp_prots: 0x0050 E164-ID: 1234 E164-ID: 4321 E164-ID: 7000 H323-ID: MJ-CME Voice Capacity Max.= Avail.= Current.= 0 14.160.110.254 1720 14.160.110.254 56804 CUBE H323-GW ENDPOINT-ID: 681D6EDC0003 VERSION: 4 AGE: 21 secs SupportsAnnexE: FALSE g_supp_prots: 0x0050 H323-ID: MJ-CUBE Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 3 BR1#sh gatek gw-type-prefix BR1#sh gatek gw-type-prefix buffer used: 219, size: 20480 GATEWAY TYPE PREFIX TABLE = Prefix: 1#* Zone CUBE master gateway list: 14.160.110.254:1720 MJ-CUBE Zone CUCM master gateway list: 14.160.110.15:1720 US_1 Prefix: 44#* Zone CME master gateway list: 14.160.115.200:1720 MJ-CME CUBE config interface GigabitEthernet0/0 ip address 14.160.110.254 255.255.255.0 duplex auto speed auto media-type rj45 h323-gateway voip interface h323-gateway voip id CUBE ipaddr 14.160.110.129 1719 h323-gateway voip h323-id MJ-CUBE h323-gateway voip tech-prefix 1# h323-gateway voip bind srcaddr 14.160.110.254 dial-peer voice 100 voip incoming called-number . dial-peer voice 200 voip destination-pattern .T session target ras CME config interface GigabitEthernet0/0 ip address 14.160.115.200 255.255.255.0 duplex auto speed auto media-type rj45 h323-gateway voip interface h323-gateway voip id CME ipaddr 14.160.110.129 1719 h323-gateway voip h323-id MJ-CME h323-gateway voip tech-prefix 44# h323-gateway voip bind srcaddr 14.160.115.200 dial-peer voice 23 voip translation-profile incoming voip-in translation-profile outgoing voip-out
Re: [OSL | CCIE_Voice] HQ-RTR in Vol2 Lab1
I think what Randall is getting at is the fact that you would put the h323-gateway voip bind source ip under the voice vlan interface. The gatekeeper source is defined under the gatekeeper with the zone local commands. Hope this helps, Jeff -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Randall Crumm Sent: Tuesday, November 30, 2010 5:40 PM To: sfuna...@cisco.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] HQ-RTR in Vol2 Lab1 You use fa 0/0/0:20 for the h323 gw and the loopback for the gk Hth Randall - Original Message - From: Satoshi Funabashi (sfunabas) [mailto:sfuna...@cisco.com] Sent: Tuesday, November 30, 2010 05:03 PM To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] HQ-RTR in Vol2 Lab1 Hello, Let me ask a question. In Vol2 Lab1, HQ-RTR act as a H323 Gateway and a Gatekeeper. But CUCM does not allow addition of H323 gateway and Gatekeeper when their IP addresses are same. When there was an entry of H323 gatway with HQ-RTR Loopback IP, I could not add Gatekeeper with same IP address. The error message was as follows: Update failed. One of the required fields on the page has the same value as an entry that already exists in the database. Please check the corresponding Find List page to verify your entry does not exist. So we need to add gateway and gatekeeper with different IP address. (In the proctor guide, gateway uses voice vlan interface and gatekeeper uses Lo0.) But because of the gatekeeper configuration, source address of H323 message will be Lo0. In this situation, when a call comes from PSTN, HQ-RTR send setup message using its source address of Lo0. As a result CUCM rejects this message and the call fails. How do I resolve this issue? Any help would be appreciated. Thanks and Regards, Satoshi Satoshi Funabashi Systems Engineer Cisco Systems G.K. Tel:81-3-6434-2824(direct) 81-3-6434-6500(group) 81-90-4050-1574(mobile) E-mail: sfuna...@cisco.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Legal Disclaimer: The information contained in this message may be privileged and confidential. It is intended to be read only by the individual or entity to whom it is addressed or by their designee. If the reader of this message is not the intended recipient, you are on notice that any distribution of this message, in any form, is strictly prohibited. If you have received this message in error, please immediately notify the sender and delete or destroy any copy of this message ___ 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 4A
Hi Rafay, Can you send the configuration from Site C? He was referring to the translation profile applied incoming on the voice-port on R3. It should look something like this: Voice-translation-rule 1 Rule 1 /^32143/ /3/ Or (I see two different called numbers below) Voice translation-rule 1 Rule 1 /^\+3432143/ /3/ Then create the profile and apply to voice-port Voice-translation-profile PSTN-IN Translate called 1 Voice-port 0/0/0:15 Translation-profile incoming PSTN-IN Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Rafay Aslam Sent: Monday, November 29, 2010 1:22 PM To: Randall Crumm Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Lab 4A Hi Attach is a complete PSNT Config. On Mon, Nov 29, 2010 at 3:12 PM, Randall Crumm randall.cr...@flextronics.com wrote: Can you send your voice translation rule/profile and how it is applied to the voice port? From: Rafay Aslam [mailto:rafayc...@gmail.com] Sent: Monday, November 29, 2010 12:11 PM To: Randall Crumm Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Lab 4A Hi I have inital configuration loaded from PSTN router, here is debug from PSTN Router, I am calling from PSTN Phone to 3214-3005, so basically I want to see if my translation pattern works on BR2 Router whyic PSTN-WAN# Nov 29 20:07:25.135: ISDN Se0/1/0:23 Q931: Ux_DLRelInd: DL_REL_IND received from L2 Nov 29 20:07:28.499: ISDN Se0/0/1:23 Q931: Ux_DLRelInd: DL_REL_IND received from L2 Nov 29 20:07:30.063: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 0xD is 0x0 0x0, Calling num 999 Nov 29 20:07:30.067: ISDN Se0/0/0:23 Q931: Sending SETUP callref = 0x0080 callID = 0x8001 switch = primary-ni interface = Network Nov 29 20:07:30.067: ISDN Se0/0/0:23 Q931: TX - SETUP pd = 8 callref = 0x0080 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Progress Ind i = 0x8583 - Origination address is non-ISDN Display i = 'Emergency Services' Calling Party Number i = 0x0080, '999' Plan:Unknown, Type:Unknown Called Party Number i = 0x80, '+3432143005' Plan:Unknown, Type:Unknown Nov 29 20:07:30.119: ISDN Se0/0/0:23 Q931: RX - CALL_PROC pd = 8 callref = 0x8080 Channel ID i = 0xA98381 Exclusive, Channel 1 Nov 29 20:07:30.131: ISDN Se0/0/0:23 Q931: RX - DISCONNECT pd = 8 callref = 0x8080 Cause i = 0x8081 - Unallocated/unassigned number Nov 29 20:07:30.135: ISDN Se0/0/0:23 Q931: TX - RELEASE pd = 8 callref = 0x0080 Nov 29 20:07:30.143: ISDN Se0/0/0:23 Q931: RX - RELEASE_COMP pd = 8 callref = 0x8080 Nov 29 20:07:30.155: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 0xD is 0x0 0x0, Calling num 999 Nov 29 20:07:30.155: ISDN Se0/0/0:23 Q931: Sending SETUP callref = 0x0081 callID = 0x8002 switch = primary-ni interface = Network Nov 29 20:07:30.155: ISDN Se0/0/0:23 Q931: TX - SETUP pd = 8 callref = 0x0081 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Progress Ind i = 0x8583 - Origination address is non-ISDN Display i = 'Emergency Services' Calling Party Number i = 0x0080, '999' Plan:Unknown, Type:Unknown Called Party Number i = 0x80, '32143005' Plan:Unknown, Type:Unknown Nov 29 20:07:30.183: ISDN Se0/0/0:23 Q931: RX - CALL_PROC pd = 8 callref = 0x8081 Channel ID i = 0xA98381 Exclusive, Channel 1 Nov 29 20:07:30.191: ISDN Se0/0/0:23 Q931: RX - DISCONNECT pd = 8 callref = 0x8081 Cause i = 0x8081 - Unallocated/unassigned number Nov 29 20:07:30.195: ISDN Se0/0/0:23 Q931: TX - RELEASE pd = 8 callref = 0x0081 Nov 29 20:07:30.203: ISDN Se0/0/0:23 Q931: RX - RELEASE_COMP pd = 8 callref = 0x8081 Nov 29 20:07:34.139: ISDN Se0/1/0:23 Q931: Ux_DLRelInd: DL_REL_IND received from L2 Nov 29 20:07:37.499: ISDN Se0/0/1:23 Q931: Ux_DLRelInd: DL_REL_IND received from L2 On Mon, Nov 29, 2010 at 12:04 PM, Randall Crumm randall.cr...@flextronics.com wrote: HI, If you can call OB you still need to make sure your config for IB is correct. You need an IB dial peer and a translation rule to strip down the called number to 3001 HTH Randall From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Rafay Aslam Sent: Monday, November 29, 2010 8:24 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Lab 4A Hi I am doing Lab 4A, I
[OSL | CCIE_Voice] I PASSED
Hi everyone, It's been a long journey, but it's finally over. Thanks for the many nights where I needed your help and you all chimed in. Thank you IPExpert for your great study materials and Vik for the final push in the 5-day bootcamp. I would recommend anyone who is about to attempt the lab and can afford it to take that bootcamp as your final push. Happy Holidays! Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Intergration of Unit express with SRST router
Hi Jason, In SRST you should point to a dial-peer that goes directly to CUE. Also add the voicemail command under telephony or voice register global that matches this pilot number. For example: Dial-peer voice 1 voip Destination-pattern 3600 Session target CUE IP address Session protocol sipv2 No vad Dtmf-relay sip-notify Codec g711ulaw You may also need a translation-profile to make the integration work. Also, for MWI to still work you would need the following: Voice service voip Sip Bind all source Loop or voice vlan Sip-ua Mwi-service ipv4:cue ip unsolicited Hope this helps, Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jason Aarons (US) Sent: Wednesday, November 24, 2010 12:32 PM To: Adam Thompson; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Intergration of Unit express with SRST router So the default dial peer for the auto attendant already points to CallManager, should I add the same autoattendant dial peers with a higher preference to the ip address of the CUE for SRST? So normal route is via CallManager/CTI Route Point but during wan failure those timeout and it goes direct to CUE? Since the phones are SIP for use with SIP SRST , I had this which doesn't account for the CUE AA when WAN is down; voice register pool 1 id network 10.1.222.0 mask 255.255.255.0 preference 1 incoming called-number dtmf-relay rtp-nte voice-class codec 1 From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Adam Thompson Sent: Tuesday, November 23, 2010 1:54 PM To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Intergration of Unit express with SRST router Take a look here: http://www.cisco.com/en/US/products/sw/voicesw/ps5520/products_configuration _example09186a0080289ef0.shtml#srst HTH -Adam On Tue, Nov 23, 2010 at 10:03 AM, Mritunjay Kumar mjs...@gmail.com wrote: Hi All, how to integrate unity express with SRST router when it is registered to cucm through JTAPI and loses connectivity with cucm ? CUE is installed in SRST router. any pointer ? Regards, Mj ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MGCP issue on BR-1 Router
All you would need to do is re-enter the bind statements if that is the case in the future. -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of cciefo...@hotmail.com Sent: Friday, November 19, 2010 10:49 AM To: ccieid1ot; ccie_voice-boun...@onlinestudylist.com; Erwan Erwan Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MGCP issue on BR-1 Router What ip is the mgcp gateway registering with? I had an issue with the same router where the ip address was the frame relay and not the loopback. This all happened even though I did everything correctly. I had to redo the mgcp config on the router to get the new ip. -Original Message- From: ccieid1ot ccieid...@gmail.com Sender: ccie_voice-boun...@onlinestudylist.com Date: Fri, 19 Nov 2010 11:52:16 To: Erwan Erwane_er...@yahoo.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MGCP issue on BR-1 Router ___ 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] ? wild card....
Try the ! wildcard instead. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of mudassar Khalid Sent: Monday, November 15, 2010 2:51 AM To: roger.car...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] ? wild card Thank you Roger. i am using same device-line CSS approach with RP. The only difference is Route pattern. Instead of 91900XXX, i am using 91900? RP. Now ? at the end of pattern is my concern. it should match any digit string with range 91 through 919.(as per admin guide) Its not the case here in my lab. only 91900.0( or any number of zeros after dot) work. if i put any non zero digit after dot(.), it doesn't match. Thanks, Mudassar _ Date: Sun, 14 Nov 2010 09:03:24 -0600 Subject: Re: [OSL | CCIE_Voice] ? wild card From: roger.car...@gmail.com To: mudas...@hotmail.com CC: ccie_voice@onlinestudylist.com Mudassar, Which wildcard did you use? 91X will block any number from 91[0 thru 9]. Actually this wouldn't let you type any 4th digit since it is the closest match when dialing a 900 numbers. To block 900 numbers; I would use the device-line CSS approach with RP 91900XXX in partition blocked assigned to the IP phones line CSS. http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/dialplan.html#wp1 150997 Regards, Roger Carpio. On Sun, Nov 14, 2010 at 8:41 AM, Mudassar Khalid mudas...@hotmail.com wrote: Hi Experts, while practicing volume 1 lab 5.9: block 91900? pattern for all phones. ? wild card has no effect if the number is not consecutive zeros after 91900 digit string. CUCM help says: The route pattern 91X? routes or blocks all numbers in the range 91 through 919. But I am not able to hit this pattern unless I dial consecutive zeros. Would anybody highlight its usage? Thanks, Mudassar ___ 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
Re: [OSL | CCIE_Voice] UCCX Script Question
For the first question - You need to either record a prompt in CUC or CUE, upload to UCCX, and then use the Play Prompt step to play to the caller. Then use the Terminate step and go to the end of the script using a label. For the second question - Once again record the prompt and upload. After that you can either create a new Play Prompt step and delete the old for the Queue Prompt Prompt variable (I assume you are using ICD) or just change the value of the Queue Prompt variable. Hope this helps, Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccieiwillb Sent: Tuesday, November 09, 2010 7:25 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] UCCX Script Question Hi Everyone, I am trying to tweak a script I put together and was curious it there is a way to play a prompt and then end the call/hangup? Also does anyone know if there is a way to not play the system message I'm sorry all of our representatives are currently assisting other callers.? I have tried a couple different solutions that I thought would work but no luck. Any help is greatly appreciated. Regards, ccieiwillb ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] GK and cube problem
Could you also send the output from the command debug voice ipipgw when you attempt a call. This would also help find the problem from a CUBE standpoint. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cristobal Priego Sent: Thursday, November 04, 2010 8:45 AM To: bruno.juniper Cc: ccie_voice Subject: Re: [OSL | CCIE_Voice] GK and cube problem do you have your Voip Dial peers configured? 2010/11/4 bruno.juniper bruno.juni...@gmail.com hello mate, I have some problem regarding GK and CUBE. HQ is gk also a cube.HQ user call 01132* go through gk. my config is below. the call didn't succeed. it show gk_rassrv_sep_arq: LRQ suspension point failed (return code = 0x4009). can anyone help me? interface FastEthernet0/0.102 description ***VOICE VLAN*** encapsulation dot1Q 102 ip address 142.102.64.254 255.255.255.0 ip helper-address 142.100.64.11 h323-gateway voip interface h323-gateway voip id VGK ipaddr 142.1.64.254 1719 h323-gateway voip h323-id CUBE h323-gateway voip bind srcaddr 142.102.64.254 ! ! gatekeeper zone local GK cisco.com 142.1.64.254 zone local VGK cisco.com zone remote BBGK cisco.com 157.1.26.253 1719 outvia VGK zone prefix BBGK 01132* no shutdown ! HQ-RTR#debug gatek ma 10 Nov 3 13:39:17.072: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Nov 3 13:39:17.076: ////GK/gk_rassrv_arq: arqp=0x4A2DE644,crv=0xB, answerCall=0 Nov 3 13:39:17.076: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Nov 3 13:39:17.076: //809F22BF0B00/809F22BF0B00/GK/gk_dns_query: No Name servers Nov 3 13:39:17.076: //809F22BF0B00/809F22BF0B00/GK/rassrv_get_addrinfo: (0113212345678) Tech-prefix match failed. Nov 3 13:39:17.076: //809F22BF0B00/809F22BF0B00/GK/rassrv_get_addrinfo: (0113212345678) Matched zone prefix 01132 and remainder 12345678 Nov 3 13:39:17.076: ////GK/gk_rassrv_get_ingress_network: returning default ingress network = 1 Nov 3 13:39:17.076: //809F22BF0B00/809F22BF0B00/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x4A04AC50 Nov 3 13:39:17.076: //809F22BF0B00/809F22BF0B00/GK/rassrv_arq_select_viazone: matched zone is GK, and z_invianamelen=0 Nov 3 13:39:17.076: //809F22BF0B00/809F22BF0B00/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x495E8FC4 Nov 3 13:39:17.076: //809F22BF0B00/809F22BF0B00/GK/rassrv_arq_select_viazone: matched zone is BBGK, and z_outvianamelen=3 Nov 3 13:39:17.076: //809F22BF0B00/809F22BF0B00/GK/rassrv_arq_select_viazone and z_outvianamep=VGK Nov 3 13:39:17.076: //809F22BF0B00/809F22BF0B00/GK/rassrv_arq_select_viazone: Received ARQ for a zone (BBGK) that has an outviazone (VGK) specified. Pick an IP-IP gateway in that viazone. Nov 3 13:39:17.076: ////GK/gk_gw_select_ipipgw_random: zonep: 0x4A297F40, tpp: 0x0, current_endpt: 1 Nov 3 13:39:17.076: ////GK/gk_gw_select_ipipgw_random: Selecting any IPIPGW. qelemp.head=0x49E0F3FC, use_count=1, current_endpt=1 Nov 3 13:39:17.076: ////GK/gk_gw_select_ipipgw_random: Gateway selection will start at the top of the linked list. use_count=1, current_endpt=0 Nov 3 13:39:17.076: ////GK/gk_gw_select_ipipgw_random: qelemp=0x49E0F3FC, loop_count=0 Nov 3 13:39:17.076: ////GK/gk_gw_select_ipipgw_random: Examining tgwp 0x49E1F0D8, g_supp_prots: 0x50 qelemp: 0x49E0F3FC, loop_count: 1 Nov 3 13:39:17.076: ////GK/gk_gw_select_ipipgw_random: Found an IPIPGW. tgwp: 0x49E1F0D8, endptsigIP: 142.102.64.254, endptrasIP: 142.102.64.254, zone: VGK Nov 3 13:39:17.076: ////GK/gk_gw_select_ipipgw_random: Selected an IPIPGW. Nov 3 13:39:17.076: //809F22BF0B00/809F22BF0B00/GK/rassrv_get_addrinfo: (0113212345678) successfully resolved IPIPGW and returning with return code 0 Nov 3 13:39:17.092: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Nov 3 13:39:17.092: ////GK/gk_rassrv_arq: arqp=0x4A2DE644,crv=0x28, answerCall=1 Nov 3 13:39:17.092: //809F22BF0B00/809F22BF0B00/GK/gk_rassrv_dep_arq: ARQ Didn't use GK_AAA_PROC Nov 3 13:39:17.108: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Nov 3 13:39:17.112: ////GK/gk_rassrv_arq: arqp=0x4A281EEC,crv=0x29, answerCall=0 Nov 3 13:39:17.112: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Nov 3 13:39:17.112: //809F22BF0B00/809F22BF0B00/GK/gk_dns_query: No Name servers Nov 3 13:39:17.112: //809F22BF0B00/809F22BF0B00/GK/rassrv_get_addrinfo: (0113212345678) Tech-prefix match failed. Nov 3 13:39:17.112: //809F22BF0B00/809F22BF0B00/GK/rassrv_get_addrinfo: (0113212345678) Matched zone prefix 01132 and remainder 12345678 Nov 3
