Re: [OSL | CCIE_Voice] Calling Search Space
The simplest way is the best way- I would set the CSS on the Device Pool under Device Mobility Related info. This is cleaner since you are assigning CSS in the fewest possible places (there are a lot more DN's and Devices than DPools). Same applies to AAR CSS- set it on the Device Pool. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Aug 3, 2013, at 10:31 PM, Karen Johnson wrote: folks, in exam , is it safe to just use CSS on DN level ? I can't think of why we need CSS for Phone level, except AAR CSS K ___ 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] NO Extension in CME-SRST
Correct. Bring out of SRST and reload. I think this is a bug with using octo lines in CME SRST in this version. Never happens on the first time going into SRST. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Aug 2, 2013, at 6:15 AM, IE Target wrote: I think that is the bug Only resolution is to relaod the router Any comments on it ?? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ 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] CCIE Collaboration officially announced
I am guessing this is a marketing decision and the technical folks feared this backlash, hence the delay in the announcement. It makes no sense whatsoever, especially as the blueprint change seems to be fairly minimal. On May 28, 2013, at 21:16, m george m.george00...@gmail.com wrote: This is quite ridiculous ! All other tracks (RS/SP) have gone through massive changes but they were retained. Even Security CCIE track has recently gone through 50% more overlap (ISE/WSA/ACS/WLC/AP what not is new) but they didn't rename it retire old one. If you look at CCIE Collaboration equipment list topics, you won't find any significant different other than TP/Jabber/InterCluster stuff which is like 15%-20% new stuff. It's so pathetic on cisco's part that they didn't value the years hardwork effort of engineers to attain Voice CCIE. I know guys who sat lab like 7 times, some even 10 times to pass. when they have finally passed this extremely tough lab, you are throwing their CCIE number in gutter by retiring a CCIE certification. Will people go for CCIE Voice lab now ? Probably NOT i bet this will be only track for which there won't be rush to complete certification. it's an extremely disappointing thing what Cisco has done. Cisco should protect investment made by tens of hundreds of engineers for years rather than giving them a retired track. For a guy who passed lab on 7th attempt recently is a Voice CCIE , will Cisco give him free vouchers 7 times to sit Collaboration CCIE now ? Morally , they should. Practically, they won't. It doesn't make sense to me . Does it make sense to anyone among you ? If so, please explain how. On Wed, May 29, 2013 at 4:08 AM, Vik Malhi vma...@ipexpert.com wrote: For my initial reaction read here: http://bit.ly/12MNK5t Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.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] CCIE Collaboration officially announced
As I said before - I would think product marketing had something to say about this. Just my opinion. Why for the last 4 years has there been a lack of Microsoft products in the exam? Are Microsoft are not relevant or it Marketing bullying the content team? At the same time there are many folks out there with a voice IE who can't spell SIP. So what do you do about those folks? Historically Cisco have trusted their recertification process as a valid check and balance. It looks like they have lost a bit of faith in the written exams as a valid means to recertification- and that is no surprise to any of us as we all know a 2nd grader could pass one of those exams (and I don't condone the means through which that is possible). When the dust has settled I will be advising all existing voice IE's two things: look at this as an extra challenge that will reap extra reward and secondly - diversify. Collaboration expertise in the very literal sense cannot be confined to a monolithic single vendor application time test that is going to occur . I would have thought a true collaboration expert would (if not now, never) be encouraged to seek skills and experience from a wider spectrum of vendors. Rant over. On May 28, 2013, at 20:14, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Yes its really frustrating what Cisco is doing to us. Ok let me tell you this. People now have invested a lot of money in pursuing their CCIE Voice that includes (Verious Workbook fees , Rack Rentals , Home Lab building , travel expenses and Lab fees attempts for whatever times) So when people achieve CCIE Voice nowadays a year or two later it would be considered old and grandfathered. Also , Cisco has released a new lab for 2 months while they are planning to abolish the whole syllabus. Why they do that to us They already make money out of everything especially lab multiple times of lab attempts per each person. CCIE Voice achievers has to send cisco request for Migration without Lab test. CCVP it was automatically migrated to CCNP Voice without any additional tests. CCNA is migrated to CCNA R/S without any additional tests. In case of Video part then I suggest whether they force CCIE Voice people to make CCNA VIDEO or CCNP Video if they will release or they make just a migration lab track that includes VIDEO stuff only for a cheaper fee something like $500. Thats same for MICROSOFT they abolished MCSE to change it to MCITP people usually just add 2 tracks to become full MCITP same when they migrate to new MCSE (Microsoft Certified Solutions Experts) there is only an upgrade track rather than taking the whole 5 tracks again. Cisco obviously has to do something like that.It's really unfair retiring the whole cisco voice totally. Guys to make the new Collaboration lab that would cost anyone over 50K to buy telepresence , X9XX routers stuff , 9971 Video Phones , TV's and etc.. Even the rack rentals would be 5 times the old voice track as the equipment would be way more expensive. Seriously , We have to agree all of us from multiple different voice study group to have a migration track to Collaboration please share your thoughts guys On 28 May 2013 18:56, Mark Holloway m...@markholloway.com wrote: Bummer, I was really hoping CCIE Voice candidates would transition to Collaboration without any additional lab exams. On May 28, 2013, at 7:08 PM, Vik Malhi vma...@ipexpert.com wrote: For my initial reaction read here: http://bit.ly/12MNK5t Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced
Correct. CCIE voice will always be certified providing they recert every two years. But there is a blemish being an IE in something that is obsolete. On May 28, 2013, at 23:14, Karen Johnson karen.johnson...@yahoo.ca wrote: all, but where is it in Cisco that said CCIE voice need to take Collaboration. if active CCIE voice keep renewing thru written, they will keep the number. From: m george m.george00...@gmail.com To: Vik Malhi vma...@ipexpert.com Cc: OSL Group ccie_voice@onlinestudylist.com Sent: Tuesday, May 28, 2013 10:16:01 PM Subject: Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced This is quite ridiculous ! All other tracks (RS/SP) have gone through massive changes but they were retained. Even Security CCIE track has recently gone through 50% more overlap (ISE/WSA/ACS/WLC/AP what not is new) but they didn't rename it retire old one. If you look at CCIE Collaboration equipment list topics, you won't find any significant different other than TP/Jabber/InterCluster stuff which is like 15%-20% new stuff. It's so pathetic on cisco's part that they didn't value the years hardwork effort of engineers to attain Voice CCIE. I know guys who sat lab like 7 times, some even 10 times to pass. when they have finally passed this extremely tough lab, you are throwing their CCIE number in gutter by retiring a CCIE certification. Will people go for CCIE Voice lab now ? Probably NOT i bet this will be only track for which there won't be rush to complete certification. it's an extremely disappointing thing what Cisco has done. Cisco should protect investment made by tens of hundreds of engineers for years rather than giving them a retired track. For a guy who passed lab on 7th attempt recently is a Voice CCIE , will Cisco give him free vouchers 7 times to sit Collaboration CCIE now ? Morally , they should. Practically, they won't. It doesn't make sense to me . Does it make sense to anyone among you ? If so, please explain how. On Wed, May 29, 2013 at 4:08 AM, Vik Malhi vma...@ipexpert.com wrote: For my initial reaction read here: http://bit.ly/12MNK5t Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.com/ Are you a CCNP or CCIE and looking for a job? Check out http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] CCIE Collaboration officially announced
For my initial reaction read here: http://bit.ly/12MNK5t Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] CCIE Voice renamed CCIE Collaboration available Nov 2013
More info to come- but we've all been waiting a long time to hear some news. People in the middle of their studies hoping to pass on the current blueprint- your countdown begins now. Vik ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CUE License File
You should see this on any of the vol 2, ILT, OWLE, 5 lab handbook base configs in c:\ftp -- Vik Malhi CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ From: nehal ahmed nehal.ah...@msn.com Date: Monday, August 13, 2012 4:54 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CUE License File Dear All, I am working on proctors lab , How to get the CUE License file , I am unable to see the file on CCX machine, Which Lab Initial COnfiguration has to be loaded to get the FTP Server settings on CCX Machine ? Reg Nehal ___ 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] 5 new labs lab 1 - cue
You don't need allow connections since it is a cti integration not sip. -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 On Jun 10, 2012, at 22:14, Randall Crumm rrcr...@yahoo.com wrote: looking at it over I think it was i did not include: voice service voip allow-connections h323 to sip etc etc etc Cheers, Randall From: Vik Malhi vma...@ipexpert.com To: Randall Crumm rrcr...@yahoo.com Cc: Online Study ccie_voice@onlinestudylist.com Sent: Saturday, June 9, 2012 8:42 PM Subject: Re: [OSL | CCIE_Voice] 5 new labs lab 1 - cue The troubleshooting methodology has to be to eliminate various items involved in the call. Change region between br2 and HQ to g711- does that work? Can you call direct to the cue pilot number from HQ/br1 or is this isolated to call forward? Can you call the cue pilot from br2 phones? Remove any location cac to eliminate an ouf bandwidth flag Does the cue have an mrgl to see the Xcoder at br2? Is the Xcoder in right dpool? Are you using RSVP? Is there above an overlapping codec in the mtp and Xcoder? -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 On Jun 9, 2012, at 12:04, Randall Crumm rrcr...@yahoo.com wrote: Hi, I am starting up again. I am trying to leave a message for branch 2 user. Branch 2 has CUE. I configured CTI PR, CTI ports,VM pilot,applied VM pilot to br2ph2 line. configured transcoder on br2 gw, registered br2 transcoder to CUCM. applied transcoder to a MRG When I call from br1 I get a fast busy Any thoughts? Cheers, 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
Re: [OSL | CCIE_Voice] Last Chance to Register for IPexpert’s Online Voice “Alchemy” Class….
Yes I can confirm I am alive and well and am at IPX. I was worried there for a while that you guys on the list had heard something so it is a relief to hear Wayne say that:-) -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ On Jun 9, 2012, at 8:56 PM, Wayne Lawson waynelawson-...@ipexpert.com wrote: Vik is definitely @ IPexpert. He's an officer and shareholder and isn't going anywhere! ;-) Regards, Wayne A. Lawson II - CCIE #5244 Founder President IPexpert, Inc., Proctor Labs, Inc., Masonic e-Institute of Technology, Inc., Platinum, Inc. Mobile: +1.810.334.1564 eFax: +1.810.454.0130 Email: wlaw...@ipexpert.com Connect @ www.WayneLawson.com :: Message sent from iPhone. On Jun 9, 2012, at 4:43 PM, donny f f.faraday...@gmail.com wrote: Hi all, Is vik not with ipx anymore? Sorry not following list for while. D On Saturday, June 9, 2012, Bill Lake whl...@gmail.com wrote: Kevin Wallace and Anthony Sequeira are both very good and helpful. The class was great and we learned ways to sharpen are skills, strategy and mindset. I think it will be well worth the time. On Thursday, June 7, 2012, Randall Crumm wrote: Hi Wayne, I was in Ken's class tonight and I have to say I think this is awesome ! Way to go. Tonight, the first night, we talked lab strategy and went over hq switch VLAN config and QoS I know from my lab experience, that this would have helped me out.I can't wait for the other 7 classes. I also like that there is bonus material from Ken. One more cool thing, the sessions are recorded, which I like because I know I won't be able to attend one of the sessions live. Bill Lake was online as well and very active on the Q and A. What did you think Bill and anyone else on there tonight? Cheers, Randall From: Wayne Lawson waynelawson-...@ipexpert.com To: OSL ccie_voice@onlinestudylist.com Sent: Wednesday, June 6, 2012 8:28 AM Subject: [OSL | CCIE_Voice] Last Chance to Register for IPexpert’s Online Voice “Alchemy” Class…. CCIE Voice Candidates - I just released a blog, and feel that's it's important for you all to check it out. I don't want this to seem like a sales and marketing pitch (although it somewhat is?, but hopefully you all see the value towards this as it pertains to your CCIE Voice 3.0 prep. I can't give any details, but I would anticipate a new Voice blueprint being announced in the very near future. If you're studying (and hoping to pass 3.0 before 4.0 is introduced) - this post will help with that goal. Last Chance to Register for IPexpert’s Online Voice “Alchemy” Class…. CCIE Voice 3.o Candidates, As most of you are aware – there are leaks that the CCIE Voice lab will be changing soon. Although there hasn’t been an official announcement – I anticipate that this announcement will come soon (possibly within the next week or so). With that being said, if you’re studying for 3.0 – your window of opportunity to pass this blueprint is shrinking! A few weeks ago we announced a new online class entitled “IPexpert’s Online CCIE Voice Alchemy Class“. This course was announced (and offered) due to, what we felt, was a need in the industry.We sought out one of the industry’s most respected Voice Instructors – Kevin Wallace – and put the course together. As you all know – IPexpert leads the industry in CCIE Voice Lab training – as we have helped certify more CCIE Voice Engineers than any company – worldwide. This course, is highly recommended by me and our team. For such a nominal cost – it’s training that will drastically improve your chances at passing this lab – in that limited window in which the 3.0 lab will be offered. Here are a few facts: Q. When does this course start? A. Thursday, June 7, 2012, 8:00 – 10:30 PM EDT Q. Who should attend this course? A. This course assumes the student has already been through at least one practice lab ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Blueprint update
The message from CIsco Live is Dont expect an announcement any time soon re: the blueprint change for CCIE-V. My interpretation is this means you have the rest of this year and fairly deep into 2013 before a blueprint change. Understandably not much information is being given but I will share my thoughts on our blog next week. If you are mid way through your studies then you have ample time- I would encourage you to not get distracted and get it done on this version. -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] ip rsvp 112 or ip rsvp 160
112 is the right answer not 160z -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 On Jun 9, 2012, at 14:16, Jason Aarons (AM) jason.aar...@dimensiondata.com wrote: Vic has some examples where he indicates to allow 4 G729s call thru rsvp to use “ip rsvp 160”, yet the SRND “shows ip rsvp 112” I can see if you want ring-in on 4 G.729 calls you would need 160. But 3 G.729 calls connected and then 1 ring-in would be 112. If I asked you to allow 4 calls in would you assume all 4 ring-in at same time and setup ip rsvp 160 (4x40) or go with ((N -1) * 24)) +40 = 112 ? Configuration Recommendation Because the initial reservation will be larger than the actual packet flow, over-provisioning the RSVP and LLQ bandwidth is required to ensure that the desired number of calls can complete. When provisioning the RSVP bandwidth value for N calls, Cisco recommends that the Nth value be the worst-case bandwidth to ensure that the Nth call gets admitted. For example: •To provision four G.729 streams: (3 * 24) + 40 = 112 kbps Reference http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/cac.html ___ 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] 5 new labs lab 1 - cue
The troubleshooting methodology has to be to eliminate various items involved in the call. Change region between br2 and HQ to g711- does that work? Can you call direct to the cue pilot number from HQ/br1 or is this isolated to call forward? Can you call the cue pilot from br2 phones? Remove any location cac to eliminate an ouf bandwidth flag Does the cue have an mrgl to see the Xcoder at br2? Is the Xcoder in right dpool? Are you using RSVP? Is there above an overlapping codec in the mtp and Xcoder? -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 On Jun 9, 2012, at 12:04, Randall Crumm rrcr...@yahoo.com wrote: Hi, I am starting up again. I am trying to leave a message for branch 2 user. Branch 2 has CUE. I configured CTI PR, CTI ports,VM pilot,applied VM pilot to br2ph2 line. configured transcoder on br2 gw, registered br2 transcoder to CUCM. applied transcoder to a MRG When I call from br1 I get a fast busy Any thoughts? Cheers, 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
Re: [OSL | CCIE_Voice] CCIE Voice new Version
Hate to speculate- let's just wait and see what happens. Either way I'm sure early 2013 we are looking at a go-live date with the new blueprint. I did talk to somebody within the program at Cisco at the UC9 beta and they mentioned they have had solid dates for an announcement for a while - so it's only just around the corner. I'll put something out on this list and our blog the moment I hear anything but if you are in the middle of your preparations then don't get distracted- you have enough time to get through on this version of the blueprint. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Jun 5, 2012, at 10:50 AM, Keith Cardoza wrote: Yes cucm 9 + video and all!!! I am already ready for it :) TC On Sun, Jun 3, 2012 at 1:16 AM, Wayne Lawson waynelawson-...@ipexpert.com wrote: Gang - I will let Vik address this, as we have already started preparing for this announcement. We *think* official details will come out at Cisco Live. Regards, Wayne A. Lawson II - CCIE #5244 Founder President IPexpert, Inc., Proctor Labs, Inc., Masonic e-Institute of Technology, Inc., Platinum, Inc. Mobile: +1.810.334.1564 eFax: +1.810.454.0130 Email: wlaw...@ipexpert.com Connect @ www.WayneLawson.com :: Message sent from iPhone. On Jun 2, 2012, at 1:41 PM, Keith Cardoza keith.cardoz...@gmail.com wrote: New ver 4 is comming in next 6 months now cisco will officially annonce the same soon. Its not at all to 2 about this bec its almost now 2 to 3 years pass in ver 3 so now they have to upgrade to ver 4 cucm 9 I just cleared my lab so did nt worried much now thanks On Fri, Jun 1, 2012 at 11:14 PM, Leslie Meade leslie.me...@lvs1.com wrote: Hmmm, I have not seen anything to say that they are going to change it.. Althou it is getting long in the tooth From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ken Wyan Sent: Friday, June 01, 2012 8:56 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CCIE Voice new Version Hi, Many people are talking about a new blueprint for CCIE Voice Lab ( should be v4). Did cisco officially announce a syllabus change for CCIE Voice Lab? Ken ___ 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] srst configuration for cbarge
I have always disabled privacy on the ephone and in the case of the privacy button- either template or ephone. And this works right away. Are you saying that disabling privacy on the ephone template without it being disabled on the ephone takes effect? Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Apr 10, 2012, at 2:27 AM, Anthony Alba wrote: I had some weirdness with the variant using auto-provision all (not auto-provision none as per the blog article) ! telephony-service srst mode auto-provision all ! In this case I expected CBarge and privacy-button to work out-of-the-box. (I have disabled single-button-barge on CUCM and configured the conference bridge to fallback to SRST) . In my testing this did not work: I had to bounce SRST mode, save the config (careful to reinput isdn bind-l3 ccm-manager), and reload the router. Now if the phones fall into SRST the ephone-template will take. Without the router reload the ephone-template seems to be ignored: i.e. privacy is on, privacy-button does not appear ephone-template 1 softkeys remote-in-use NewCall CBarge privacy off privacy-button Does it work for you folks immediately? ___ 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] CCIE voice lab Hardware list
\tx5040\tx5760\tx6480\tx7200\tx7920\tx8640\ql\qnatural\pardirnatural \b\fs42 \cf0 SiteC-RTR\ \pard\tx720\tx1440\tx2160\tx2880\tx3600\tx4320\tx5040\tx5760\tx6480\tx7200\tx7920\tx8640\ql\qnatural\pardirnatural \b0\fs24 \cf0 \ \ SiteC-RTR#sh inv\ NAME: 2811 chassis, DESCR: 2811 chassis\ PID: CISCO2811 , VID: V04 , SN: FTX1123F09U\ \ NAME: Two port E1 voice interface daughtercard on Slot 0 SubSlot 0, DESCR: Two port E1 voice interface daughtercard\ PID: VWIC-2MFT-E1 , VID: V01, SN: 35897212 \ \ NAME: One port T1 voice interface daughtercard on Slot 0 SubSlot 1, DESCR: One port T1 voice interface daughtercard\ PID: VWIC-1MFT-T1= , VID: 1.0, SN: 32942346 \ \ NAME: 4 Port FE Switch on Slot 0 SubSlot 3, DESCR: 4 Port FE Switch\ PID: HWIC-4ESW , VID: V01 , SN: FOC12340M3R\ \ NAME: WIC/VIC/HWIC 3 Power Daughter Card, DESCR: 4-Port HWIC-ESW Power Daughter Card\ PID: ILPM-4, VID: V01 , SN: FOC12334ES5\ \ NAME: PVDMII DSP SIMM with one DSP on Slot 0 SubSlot 4, DESCR: PVDMII DSP SIMM with one DSP\ PID: PVDM2-16 , VID: V01 , SN: FOC11194QV0\ \ NAME: PVDMII DSP SIMM with one DSP on Slot 0 SubSlot 5, DESCR: PVDMII DSP SIMM with one DSP\ PID: PVDM2-16 , VID: V01 , SN: FOC124758M5\ \ NAME: AIM Service Engine 0, DESCR: AIM Service Engine\ PID: AIM-CUE , VID: V02 , SN: FOC111757F2\ \ \ \ SiteC-RTR#sh cdp n\ Capability Codes: R - Router, T - Trans Bridge, B - Source Route Bridge\ S - Switch, H - Host, I - IGMP, r - Repeater\ \ Device IDLocal Intrfce HoldtmeCapability Platform Port ID\ SiteA-RTRSer 0/1/0:0.1 165 R S I 2811 Ser 0/0/1:0.2\ SC-PH1-7962 Fas 0/3/3 160 H IP Phone Port 1\ \pard\tx720\tx1440\tx2160\tx2880\tx3600\tx4320\tx5040\tx5760\tx6480\tx7200\tx7920\tx8640\ql\qnatural\pardirnatural \cf0 SC-PH2-7962 Fas 0/3/2 147 H IP Phone Port 1\ \pard\tx720\tx1440\tx2160\tx2880\tx3600\tx4320\tx5040\tx5760\tx6480\tx7200\tx7920\tx8640\ql\qnatural\pardirnatural \cf0 \ \ \ \ \ \pard\tx720\tx1440\tx2160\tx2880\tx3600\tx4320\tx5040\tx5760\tx6480\tx7200\tx7920\tx8640\ql\qnatural\pardirnatural \b\fs42 \cf0 PSTN/FR Switch\ \pard\tx720\tx1440\tx2160\tx2880\tx3600\tx4320\tx5040\tx5760\tx6480\tx7200\tx7920\tx8640\ql\qnatural\pardirnatural \b0\fs24 \cf0 \ \ PSTN-WAN#sh inv\ NAME: 2811 chassis, DESCR: 2811 chassis\ PID: CISCO2811 , VID: V04 , SN: FTX1123F09X\ \ NAME: Two port T1 voice interface daughtercard on Slot 0 SubSlot 0, DESCR: Two port T1 voice interface daughtercard\ PID: VWIC-2MFT-T1 , VID: V01, SN: 35869733 \ \ NAME: One port T1 voice interface daughtercard on Slot 0 SubSlot 1, DESCR: One port T1 voice interface daughtercard\ PID: VWIC-1MFT-T1= , VID: 1.0, SN: 22772395 \ \ NAME: Two port E1 voice interface daughtercard on Slot 0 SubSlot 2, DESCR: Two port E1 voice interface daughtercard\ PID: VWIC-2MFT-E1 , VID: V01, SN: 35897045 \ \ NAME: Two port T1 voice interface daughtercard on Slot 0 SubSlot 3, DESCR: Two port T1 voice interface daughtercard\ PID: VWIC-2MFT-T1 , VID: V01, SN: 35893236 \ \ NAME: PVDMII DSP SIMM with one DSP on Slot 0 SubSlot 4, DESCR: PVDMII DSP SIMM with one DSP\ PID: PVDM2-16 , VID: V01 , SN: FOC11194QUR\ \ } Vik Malhi – CCIE #13890Managing Partner - IPexpert, Inc.Telephone: +1.810.326.1444 ext 420Fax: +1.810.454.0130Mailto:vma...@ipexpert.com On Apr 10, 2012, at 9:09 AM, Ashutosh Dubey wrote:Hi Guys, I am planning to set up IP Expert CCIE voice lab at home. Does anybody have the list of part numbers and there respective quantityI should be purchasing. Thank you very much for your assistance. Regards, Ashutosh___For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comAre you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] srst configuration for cbarge
have a quick look at this if you are an expert cbarger Thanks Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.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] Vik's blog on CFUR - manipulate XML display of called number on calling device.
Juan- you are correct- the RP will not be used to update the caller's display when Called Party Transformation patterns are used. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 19, 2012, at 2:50 PM, Juan Lopez wrote: Baktha, I read your response on Vik's blog for CFUR. I try to manipulate the called number on the XML display of the caller's phone so that it would look like an internal call - by setting a called party transformation at the RP used by CFUR - like you suggest. Only thing is that this does not work whenever you have called party transformations at the CUCM egress gateway - these even do overwrite the XML display for the called number on the calling device according my tests. So how does this work taking your response into consideration, where you say to work with called party transformations on the egress gateway? Does this work for you? - if so, would you want to share how you setup that part of the dialplan? cheers, Juan Op 19 maart 2012 18:47 schreef Baktha Muralidharan muralic...@gmail.com het volgende: Steve Congratulations!! Enjoy the well-deserved break! /Baktha ___ 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] ip rsvp bandwidh issues
Does deb ip rsvp sign show anything? Does deb sccp events show anything? (there will be a keep alive - anything besides this?) Normal problems with RSVP 1. the CODEC within the dspfarm is not g729r8 2. the IOS Enhanced MTP is not in the appropriate DPool. 3. the MRGLMRGMTP is incorrect or has not been assigned. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 14, 2012, at 2:49 PM, Emanuel Damasceno wrote: Hello experts. I am experiencing a quite annoying problem here on my Lab. After I configure everything and start testing, I see my HQ and BR1 phones are always sending to AAR (Network Congestion. Retouting) on the first call. I am adding ip rsvp bandwidth on the serial interface and ip rsvp bandwidh xxx on the sub-interface. If I remove ip rsvp bandwidh from serial, it stops working. Now with both commands applied it is not working... What is the correct order to troubleshoot this? Does that button Resync Bandwidth on CUCM help us in any way? Thanks Emanuel Damasceno CCNP Voice ___ 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] QoS 3750
I would 100% recommend running auto qos on an UNUSED port for the base config it adds in global config (e.g. cos-dscp maps) - but MANUALLY add port specific commands if required (e.g srr, pr out, service-policy). Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 10, 2012, at 11:55 AM, Emanuel Damasceno wrote: I found this to be very interesting... I remember some people asking about the priority-queue out command in earlier posts. As I am fine tuning my QoS skills, I found this here on the Command Reference Guide for Catalyst 3750: Usage Guidelines When you configure the priority-queue out command, the shaped round robin (SRR) weight ratios are affected because there is one fewer queue participating in SRR. This means that weight1 in the srr-queue bandwidth shape or the srr-queue bandwidth shape interface configuration command is ignored (not used in the ratio calculation). The expedite queue is a priority queue, and it is serviced until empty before the other queues are serviced. Follow these guidelines when the expedite queue is enabled or the egress queues are serviced based on their SRR weights: • If the egress expedite queue is enabled, it overrides the SRR shaped and shared weights for queue 1. • If the egress expedite queue is disabled and the SRR shaped and shared weights are configured, the shaped mode overrides the shared mode for queue 1, and SRR services this queue in shaped mode. • If the egress expedite queue is disabled and the SRR shaped weights are not configured, SRR services the queue in shared mode. So, watch out for auto qos, because it adds priority-queue out on the port you applied auto qos... Emanuel Damasceno CCNP Voice On Sat, Mar 10, 2012 at 4:44 PM, Emanuel Damasceno aedamasc...@gmail.com wrote: Hello Expert, I am curious... When we issue auto qos command on a port, does it mess around with queue-set 1 or queue-set 2? I was told by a friend that when we do auto qos it messes around with queue-set 1, but I also see equal configs on queue-set 2. Can anybody share some thoughts? Thanks Emanuel Damasceno CCNP Voice ___ 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] responding to emails on this list
As a reminder- please read my email below. We do ban anybody who violates the NDA on purpose with ulterior motives but new accounts keep appearing. Please don't respond to the emails. Many Thanks! Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Feb 6, 2012, at 11:45 AM, Vik Malhi wrote: Please don't respond to the folks trying to advertise on this list. There are certain companies that have produced a series of labs (evidently one thru six...and counting) that are trying to hijack whatever forum they can to sell their products. I think Cisco are wise to what is going on and will continue to make changes to labs to protect the integrity of the lab- this is one of the purposes of the troubleshooting aspect of the lab. We can all have the same question but the answer for each and every one of us can be different. I don't want to preach- but regardless of your opinion- if you do feel the need to response please do this unicast and not copy the list on any responses. By the way- I'm offering 45 days for the over/under for lab #7 for any takers:-) Thanks! Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.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] Route Pattern DDI
In Route pattern configuration page , under DDI what's the difference between the selectebale options None No Digits ? Let me answer this question slightly differently to how it has been asked. If you select the default of None in the Route List then the RP manipulation is used. If you select NoDigits in the Route List then the RP manipulation is not used since the RL has a non-default value. So if you dialed 9911 and on the RP: 9.911 you have DDI- predot. This means the caller see's To 911 on his/her phone. If you have Nodigits on the RL the gateway see's 9911. If you have None on the RL then the gateway see's 911 (since RP manipulation is used). Also why PREDOT DDI is not available in Route group DDI box ? (we have to select NANP DDI even if we don't use @ patterns) This is because a RL can be used for multiple installed numbering plans when Route Filters are used. Saves you repetition of RL's. The benefit can't be seen when there is only 1 Numbering plan installed... Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 13, 2012, at 6:23 AM, Ken Wyan wrote: In Route pattern configuration page , under DDI what's the difference between the selectebale options None No Digits ? Also why PREDOT DDI is not available in Route group DDI box ? (we have to select NANP DDI even if we don't use @ patterns) Thanks in Advance ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Route Pattern DDI
H.see my response and let me know if this makes sense. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 13, 2012, at 4:03 PM, Mohammed Al Baqari wrote: No digits means that all digits will be striped. None means that all digits will be forwarded without any modification at RP level. Regards, Mohammed Al Baqari From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ken Wyan Sent: Tuesday, March 13, 2012 5:23 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Route Pattern DDI In Route pattern configuration page , under DDI what's the difference between the selectebale options None No Digits ? Also why PREDOT DDI is not available in Route group DDI box ? (we have to select NANP DDI even if we don't use @ patterns) Thanks in Advance ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 73, Issue 53
For the record- the 45 days I mentioned earlier in this thread was just a joke- I have no idea about lab 7. All I was doing was I guessing when people would start talking about lab 7 on this mailing list. I was asking the group would it be more or less than 45 days. As it turns out- it was 37 days as Randall points out- maybe I should give up my day job and become a professional gambler? Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 13, 2012, at 7:08 PM, Rrcrumm wrote: 37 days lol Sent from my iPhone On Mar 13, 2012, at 6:33 PM, Ramon De La Cruz ramon.delac...@sbcglobal.net wrote: Hi Vik, How's it going? Your email caught my eye...and interested in what the 45 days for the over/under for lab #7 for any takers is about. Thanks, Ramon De La Cruz From: ccie_voice-requ...@onlinestudylist.com ccie_voice-requ...@onlinestudylist.com To: ccie_voice@onlinestudylist.com Sent: Tue, March 13, 2012 6:43:50 PM Subject: CCIE_Voice Digest, Vol 73, Issue 53 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: responding to emails on this list (Vik Malhi) 2. Re: Route Pattern DDI (Vik Malhi) 3. Re: Route Pattern DDI (Vik Malhi) -- Message: 1 Date: Tue, 13 Mar 2012 16:11:50 -0700 From: Vik Malhi vma...@ipexpert.com To: OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] responding to emails on this list Message-ID: 8364df8a-124b-4aa6-89e3-b93cd652c...@ipexpert.com Content-Type: text/plain; charset=windows-1252 As a reminder- please read my email below. We do ban anybody who violates the NDA on purpose with ulterior motives but new accounts keep appearing. Please don't respond to the emails. Many Thanks! Vik Malhi ? CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Feb 6, 2012, at 11:45 AM, Vik Malhi wrote: Please don't respond to the folks trying to advertise on this list. There are certain companies that have produced a series of labs (evidently one thru six...and counting) that are trying to hijack whatever forum they can to sell their products. I think Cisco are wise to what is going on and will continue to make changes to labs to protect the integrity of the lab- this is one of the purposes of the troubleshooting aspect of the lab. We can all have the same question but the answer for each and every one of us can be different. I don't want to preach- but regardless of your opinion- if you do feel the need to response please do this unicast and not copy the list on any responses. By the way- I'm offering 45 days for the over/under for lab #7 for any takers:-) Thanks! Vik Malhi ? CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20120313/7ee18fb7/attachment-0001.html -- Message: 2 Date: Tue, 13 Mar 2012 16:30:08 -0700 From: Vik Malhi vma...@ipexpert.com To: Ken Wyan kew...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Route Pattern DDI Message-ID: c9982f50-d23d-43b5-bb5e-4b47b940a...@ipexpert.com Content-Type: text/plain; charset=windows-1252 In Route pattern configuration page , under DDI what's the difference between the selectebale options None No Digits ? Let me answer this question slightly differently to how it has been asked. If you select the default of None in the Route List then the RP manipulation is used. If you select NoDigits in the Route List then the RP manipulation is not used since the RL has a non-default value. So if you dialed 9911 and on the RP: 9.911 you have DDI- predot. This means the caller see's To 911 on his/her phone. If you have Nodigits on the RL the gateway see's 9911. If you have None on the RL then the gateway see's 911 (since RP manipulation is used). Also why PREDOT DDI is not available in Route group DDI box ? (we have to select NANP DDI even if we don't use @ patterns) This is because a RL can be used for multiple installed numbering plans
Re: [OSL | CCIE_Voice] UnifiedFx PhoneView does not work with cme
Can you confirm the version of IOS on the BR2. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 12, 2012, at 6:03 AM, The Masterplan wrote: Hi, I'm interested if anyone had the following issue. I was solving workbook 2 lab 1 and I was trying to manage branch2 phones registered in cme at branch 2 site with PhoneView. I pasted the lines mentioned in the pdf into the router config, but nothingafter I tested that the group has connectivity with cme and click add no phone appears in the interface. UCM works perfectly. I have Windows 7 x64 and .NET framework ver. 4. ___ 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] HQ call SC and transfer to CUE
Make sure you are not using ANY voice-class codec on the dial-peer from GK and the dial-peer to CUE. Also make sure you allow H323 to SIP connections. If this does not help send me the entire config. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 12, 2012, at 7:45 AM, mercy forall wrote: Hi all tried to call cue form HQ , i can not give me dissconect , the call use codeck g729 , i install transcoder 3 session in site c voice mail work in sc and from pstn , but if the call come through GK disconnect , give me disconnect code 47 i review all configuration , and also my frind review it , no issue in configuratin , i dont know why ? is this hardware issue , or miss conf debug ccsip mess 2-R3# Mar 12 07:18:41.612: //-1//SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:4220@177.3.11.2:5060 SIP/2.0 Via: SIP/2.0/UDP 177.3.11.1:5060;branch=z9hG4bK6974 Remote-Party-ID: HQPH2 sip:2002@177.3.11.1;party=calling;screen=yes;privacy=off From: HQPH2 sip:2002@177.3.11.1;tag=5DD76C-1051 To: sip:4220@177.3.11.2 Date: Mon, 12 Mar 2012 07:18:41 GMT Call-ID: 6DB359A6-6B4A11E1-804CE032-8E5D7E0@177.3.11.1 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 2160937878-1352913397-469769730-16575 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Timestamp: 1331536721 Contact: si BR2-R3#p:2002@177.3.11.1:5060 Expires: 180 Allow-Events: telephone-event Content-Length: 0 Mar 12 07:18:41.628: //-1//SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 177.3.11.1:5060;branch=z9hG4bK6974 To: sip:4220@177.3.11.2;tag=cue7b03fb81 From: HQPH2 sip:2002@177.3.11.1;tag=5DD76C-1051 Call-ID: 6DB359A6-6B4A11E1-804CE032-8E5D7E0@177.3.11.1 CSeq: 101 INVITE Content-Length: 0 Timestamp: 1331536721 Contact: sip:4220@177.3.11.2:5060 Mar 12 07:18:41.628: //-1//SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 177.3.11.1:5060;branch=z9hG4bK6974 To: sip:4220@177.3.11.2;tag=cue7b03fb81 From: HQPH2 sip:2002@177.3.11.1;tag=5DD76C-1051 Call-ID: 6DB359A6-6B4A11E1-804CE032-8E5D7E0@177.3.11.1 CSeq: 101 INVITE Content-Length: 0 Contact: sip:4220@177.3.11.2:5060 Mar 12 07:18:41.632: //-1//SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 Ok Via: SIP/2.0/UDP 177.3.11.1:5060;branch=z9hG4bK6974 To: sip:4220@177.3.11.2;tag=cue7b03fb81 From: HQPH2 sip:2002@177.3.11.1;tag=5DD76C-1051 Call-ID: 6DB359A6-6B4A11E1-804CE032-8E5D7E0@177.3.11.1 CSeq: 101 INVITE Content-Length: 174 Contact: sip:4220@177.3.11.2:5060 Content-Type: application/sdp Call-Info: sip:177.3.11.2:5060;method=NOTIFY;Event=telephone-event;Duration=2000 Allow-Events: telephone-event v=0 o=CiscoSystemsSIP-Workflow-App-UserAgent 1350 1350 IN IP4 177.3.11.2 s=SIP Call c=IN IP4 177.3.11.2 BR2-R3# t=0 0 m=audio 16904 RTP/AVP 0 a=rtpmap:0 pcmu/8000 a=ptime:20 Mar 12 07:18:41.636: //-1//SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:4220@177.3.11.2:5060 SIP/2.0 Via: SIP/2.0/UDP 177.3.11.1:5060;branch=z9hG4bK6974 Remote-Party-ID: HQPH2 sip:2002@177.3.11.1;party=calling;screen=yes;privacy=off From: HQPH2 sip:2002@177.3.11.1;tag=5DD76C-1051 To: sip:4220@177.3.11.2 Date: Mon, 12 Mar 2012 07:18:41 GMT Call-ID: 6DB359A6-6B4A11E1-804CE032-8E5D7E0@177.3.11.1 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 2160937878-1352913397-469769730-16575 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Timestamp: 1331536721 Contact: sip:2002@177.3.11.1:5060 Expires: 180 Allow-Events: telephone-event Content-Length: 0 Mar 12 07:18:41.644: //-1//SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 Ok Via: SIP/2.0/UDP 177.3.11.1:5060;branch=z9hG4bK6974 To: sip:4220@177.3.11.2;tag=cue7b03fb81 From: HQPH2 sip:2002@177.3.11.1;tag=5DD76C-1051 Call-ID: 6DB359A6-6B4A11E1-804CE032-8E5D7E0@177.3.11.1 CSeq: 101 INVITE Content-Length: 174 Contact: sip:4220@177.3.11.2:5060 Content-Type: application/sdp Call-Info: sip:177.3.11.2:5060;method=NOTIFY;Event=telephone-event;Duration=2000 Allow-Events: telephone-event v=0 o=CiscoSystemsSIP-Workflow-App-UserAgent 1350 1350 IN IP4 177.3.11.2 s=SIP Call c=IN IP4 177.3.11.2 t=0 0 m=audio 16904 RTP/AVP 0 a=rtpmap:0 pcmu/8000 a=ptime:20 Mar 12 07:18:41.980: //-1//SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 Ok Via: SIP/2.0/UDP 177.3.11.1:5060;branch=z9hG4bK6974 To: sip:4220@177.3.11.2;tag=cue7b03fb81 From: HQPH2 sip:2002@177.3.11.1;tag=5DD76C-1051 Call-ID: 6DB359A6-6B4A11E1
Re: [OSL | CCIE_Voice] Called-# representation on Calling-phone
You have to remember that the digit manipulation is bound to the ROUTE LIST and not Route Group. So the SLRG benefits can still be seen since- you would just need to create a few more Route Lists (which isn't a large overhead) each containing SLRG. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 12, 2012, at 4:56 AM, Steven wrote: Hi List, i was wondering if we need to manipulate the Called # on the phne display in the way a user doesn't notice anything. For example: I have RP 9.1XXX with a RL assigned that includes only the SLRG. PreDot is applied to this RP. When a user dials 91000 the Pattern matches and the user-display changes to 1000 Is this the correct way or should it display 91000 instead of 1000 If the 9 is needed to be displayed i would make my SLRG with PreDot by default. I'm struggling with this because i think i would waste a lot of the benefits a SLRG brings. Thanks and Regards, Steven ___ 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] call dropped after change SIP trunk MTP Preferred Originating Codec
You must ensure that the MRGL of the SIP Trunk now points to an IOS Enhanced MTP which contains codec g729r8 as the codec (within the IOS). When using the g711u codec you can use the UCM built-in MTP or the IOS Enhanced MTP (which would have codec g711u as the chosen codec). Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 12, 2012, at 2:14 AM, Guoming Zhang wrote: I am doing Vol 1 lab 5c, 5.2. Everything works fine and I call make call from HQ or BR1 to BR2 via SIP trunk. However, if I just changed the MTP preferred Originating Codec from g711ulaw to G729 or G729b, call will immediately drop if I call from BR1 to BR2 SCCP phone, but it does not matter if I call from BR1 to BR2 SIP phones. However if I make call from BR2 SCCP to BR1, it works fine. I have enabled debug ccsip all, below is the output. Can anyone tell me what is wrong here? ___ 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] QoS Question
Ok- good answer. The one I'm questioning is this: in English when would traffic being placed into egress Q4T3 be dropped? Q4T2 Traffic will be dropped if it exceeds 67% of X unless it could grab 33% of X from the common pool, in this case Q4T3 traffic will be dropped when it reaches X. If this is the case what is the purpose of the maximum threshold (M in my PDF). Is this totally unused? This is maximum memory that this queue can have before packets are dropped to quote the 3750 config guide. Q4T1 and Q4T2 are not allowed to expand their buffers to M (400%) since their threshold values are set to 20% and 50% respectively. How about Q4T3? The Q4 buffer could expand to M to avoid dropping traffic placed into Q4T3 (if common pool BW is available!). Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 8, 2012, at 4:23 AM, Amine Samaha wrote: Questions for the group (use the PDF to help you). in English when would traffic being placed into egress Q4T1 be dropped? Q4T1 Traffic will be dropped if it exceeds 20% of X in English when would traffic being placed into egress Q4T2 be dropped? Q4T2 Traffic will be dropped if it exceeds 50% of X in English when would traffic being placed into egress Q4T3 be dropped? Q4T2 Traffic will be dropped if it exceeds 67% of X unless it could grab 33% of X from the common pool, in this case Q4T3 traffic will be dropped when it reaches X. I hope this is correct!! Subject: Re: [OSL | CCIE_Voice] QoS Question From: vma...@ipexpert.com Date: Wed, 7 Mar 2012 11:39:40 -0800 CC: ccie_voice@onlinestudylist.com To: amine_sam...@hotmail.com The speed of the interface (100Mbps) is NOT the buffer size but rather the bandwidth. You configure the breakdown of this within each interface for example: SiteA-Switch(config)#int f1/0/1 SiteA-Switch(config-if)#srr bandwidth share 25 25 25 25 The output buffer is when there is congestion for traffic outbound in a specific direction e.g. a gateway/router, server, phone. If the switch's sending rate is greater than the reciever can handle we need to buffer the traffic. Do we want to wait for the buffer to become full and then just drop everything (Tail Drop)? No. We want to apply weights so that lower priority traffic is dropped before the buffers become full (WTD- congestion avoidance). So with your example: f1/0/2 bandwidth: 100M queue: 4 buffer: 54 threshold1: 20 threshold2: 50 reserved: 67 maximum: 400 Q4 has 54% of the buffers assigned to f1/0/2. This has got nothing to do with the speed of the interface. The buffer size looks like a small value - I think 2MB per 4 ports (I don't know if this is published). Please see the attached PDF for a graphic illustration of this interfaces' egress WTD. Questions for the group (use the PDF to help you). in English when would traffic being placed into egress Q4T1 be dropped? in English when would traffic being placed into egress Q4T2 be dropped? in English when would traffic being placed into egress Q4T3 be dropped? Vik Malhi � CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 6, 2012, at 1:08 PM, Amine Samaha wrote: Hi Vik, In reference to the blog articles you've mentioned below, kindly, i need to clarify two points related to egress queuing: 1) Let us assume the example below: f1/0/2 bandwidth: 100M queue: 4 buffer: 54 threshold1: 20 threshold2: 50 reserved: 67 maximum: 400 Is it true in this case that Q4 T1 = 20% of 54 = 10.8M and Q4 T3 = 100% of (67% of 54) = 36.18M and maximum (max BW allowed to be grabbed from the common pool during congestion) is = 400% of 54 = 216M 2) if maximum is set to 100 i could understand that Q4 will not be allowed to borrow any bandwidth from the common pool during congestion. Is this correct Thanks a lot, from: vma...@ipexpert.com Date: Mon, 5 Mar 2012 14:56:33 -0800 To: k...@rogers-mail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] QoS Question We are using the T1 and T2 values on the egress side in a very different way to the way we use T1/T2 on the ingress side. We are trying to expand our buffers dynamically to prevent the frame from being dropped. How we do this is by not reserving all of our memory per port (in our case we reserved 92%) and contributing to a common pool which can be used for the interfaces that are congested and need the extra buffer space. The dynamic nature of the reserved/max threshold is more flexible that the more regimented method you have described- which may be good for some ports but not others (and you only get two shots since there are only two queue sets). Vik Malhi � CCIE #13890 Managing Partner
Re: [OSL | CCIE_Voice] VPIM
Is this your CUC or CUE? The demo license on CUC does not allow you to add VPIM locations. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 8, 2012, at 4:51 PM, Cisco Nut wrote: Hi I am running a Demo license on my CUE server, when I add VPIM location it gives me an error that VPIM is a license feature, Please let me kow how you guys are working on VPIM in your home labs. Please see below exact error I get when I tried adding VPIM location. Regards Status The requested operation would result in a license violation. Unable to create VPIM Location ___ 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] CUCME and home rack
CME ver 7.0(1) This can either be 12.4(20)T or 12.4(22)T Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 8, 2012, at 4:55 AM, Gregory Wenzel wrote: Could someone be so kind to tell me what version cme is running in BR2? TIA -- Greg Wenzel, CCNP Voice ___ 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] UC SIP integration question
No- you don't need authentication. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 8, 2012, at 3:40 AM, Farkas Péter wrote: Hi, Creating a CUCM-UC SIP integration do we need to configure SIP authentication and registration under port group configuration page of UC? It seems to be working w/o but DSG for W2Lab7 fills these items, as well. Peter ___ 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] Mobile Voice Access
Welcome Mathew. Juan- sporadic in all 7.x. Sometimes works, sometimes doesn't. My advice- don't go anywhere near partial matching for the lifetime of your CCIE-V Lab prep. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 6, 2012, at 9:26 PM, Juan Lopez wrote: is this a general issue? I did not have this kind of problem: RP set to +e164 and partial match did not give any problems in my case... Used UCM version is 7.0.1.11000-2 thx! Juan 2012/3/6 Mathew Miller miller.mat...@gmail.com Thanks much Vik! I have confirmed that if I have a 10 digit RD and get 7 digit ANI and set to Partial Match that every time I experience this behavior with the annunciator. When I changed it to full match with a 7 digit RD it fixed my problem. On Tue, Mar 6, 2012 at 11:08 AM, Vik Malhi vma...@ipexpert.com wrote: I'm guessing you are using partial matching of the ANI versus the RD. Make it a complete match- in other words whatever the ANI is (from the PSTN in the Q931 debug) make this the RD number. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 6, 2012, at 8:29 AM, Mathew Miller wrote: Hello All, I have setup mobile voice access for use to be able to dial extensions using enterprise access in my home lab. 100 times out of 100 it works just peachy… I setup my dial-peer on my router, I setup the IVR service on the router, I setup my RDP and RD, along with my Mobile Voice Access number and set my Service Parameters. In my lunch dates I set this up exactly like I have done 100 times at home and everything seems to work fine until I try to dial the the extension I want to call and I get the Annunciator telling me my call can't be completed as dialed. I have checked my re-routing CSS and all the steps in setup and have access to internal extensions so I don't know what I am doing wrong. I have tried to create the issue in my home lab and cant seem to do it. It work EVERY time in my home lab. Can anyone think of something I may be overlooking? ___ 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] Call Routing - Calling Type Presentations
Your point is valid- the Telco would normally manipulate the Calling Number Types accordingly to prevent the situation you have described. For the purposes of the lab you should only really test calls from the PSTN phone into HQ/BR1/BR2. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 6, 2012, at 7:27 AM, Jason Murray wrote: I'm going through Vol 2 Lab4 Task 2, Call Routing. I get the concepts its trying to convey and I am beginning to think that this is a good way to cover alot of different routing questions that could be thrown at you. But I am finding a flaw in the part about setting all calling numbers out as National. In the question it isnt a requirement but if you do configure your calling number presentations by using the gw and calling party number transformation patterns and set it as National, any time you call a BR2 phone the incoming number comes in as National so on the gateway settings a branch 2 all numbers that come in as National gets prefixed with +34. Well HQ-BR1 sent out the gateway as National back into BR2 gets shown as +3412123945002. Whats the best way to fix that? I know normally its an internal number so the user should just be dialing 3002 instead the full international number. Buts lets say they do dial it that way. I tried to manipulate the Translation Pattern for international to set the type as International but I guess the gw CPTP overrides those settings. Would you just not do any gw CPTPs and just set those individually then? Just curious. Like I said the question doesnt address it, just want to know for my own benifit. Thanks Jason ___ 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] CCIE Numbers ?
Only Cisco knows. One thing for sure is that the pass rate was very very high (based on numbers) and this cannot (in my opinion) be without suspicion. If the numbers have flattened I would think they have found and addressed the source of the problem. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 4, 2012, at 10:45 PM, Ken Wyan wrote: CCIE Numbers reached 30,000 in last September. In January they were giving 34k numbers . Now in March still 34k numbers. Seems pass-rate has dropped in year 2012 ? Lab or Grading seems got tougher this year. ___ 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] cBarge during SRST
You have to disable privacy at the ephone level (in IOS 12.4x). Disabling privacy at the telephony-service level and template level does not work. Therefore you must have srst mode auto-prov all in order to preserve cbarge. So I would expect that if you are required to preserve cBarge, the words do not pre-define any ephone or ephone-dn to not restrict the learned ephones/dn's from showing up in the running config. The problem I have with not creating any pre-defined ephones/dn's is that you must have a pre-defined ephone-dn for the conference ad-hoc so I'm not sure if this question is possible. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 4, 2012, at 2:50 AM, Vega Wong wrote: Hi All, Let say we need to make sure the same call feature remains during SRST, and there is a share line with cbarge during normal CUCM operation. However, if the requirement is Do not pre-define any ephone or ephone-dn in running-config how would you interupt this? The simplest way is to use srst mode auto-provision none. The issue with this command is that cBarge during SRST would not work. So if we use srst mode auto-provision all, the configure in running-config will be learnt during SRST. would that still consider as pre-define? Cheers ___ 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] CUCME time format
I don't think so. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 4, 2012, at 9:33 PM, Ken Wyan wrote: Can we change date display seperator ( / slash , - dash , . dot ) in CUCME call-manager fallback similar to CUCM. Is there any command except date-format , time-format in CME for this? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] QoS Question
We are using the T1 and T2 values on the egress side in a very different way to the way we use T1/T2 on the ingress side. We are trying to expand our buffers dynamically to prevent the frame from being dropped. How we do this is by not reserving all of our memory per port (in our case we reserved 92%) and contributing to a common pool which can be used for the interfaces that are congested and need the extra buffer space. The dynamic nature of the reserved/max threshold is more flexible that the more regimented method you have described- which may be good for some ports but not others (and you only get two shots since there are only two queue sets). Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 5, 2012, at 1:44 PM, Kyle Rogers wrote: Vik, Thanks for the explanation, that answered most of my questions and helped quite a bit. My only other question is why someone would carve out 10% of the buffers for a queue, but reserve an amount other than 100%. For example, if I set the Reserved Bandwidth to 80, why wouldn't I just set the buffer setting to 8 instead? The only explanation I can come up with is that I can only use whole percentages in the buffer statement and can't put 8.5%, but if I put 10% buffers and 85% reserved, I can reserve 8.5% of the buffers. Is that the reason or am I missing a piece of the puzzle? I apologize for asking so many questions but I'm sort of at an impass in my studies until I get a firm grasp on this. I will definitely check out the blog. Thanks, Kyle On Mon, Mar 5, 2012 at 3:35 PM, Vik Malhi vma...@ipexpert.com wrote: Answers inline. For more info please read my 3 part blog on the Catalyst 3750: http://blog.ipexpert.com/tags/3750-qos/ Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 5, 2012, at 11:26 AM, Kyle Rogers wrote: QoS is probably the area that I have the most difficulty with - especially LAN QoS. I have some general questions. let's use the following sample config: mls qos queue-set output 1 buffers 10 10 26 54 mls qos queue-set output 1 threshold 2 138 138 92 400 You have only showed queue set 1 - we shall assume that the interface is assigned to queue set 1 but you must check the interface. Let's say this is applied to a 100 Mbps interface So if I understand this correctly: Queue 1 = 10% of interface bandwidth is reserved (10 Mbps) Queue 2 = 10% of interface bandwidth is reserved (10 Mbps) Queue 3 = 26% of interface bandwidth is reserved (26 Mbps) Queue 3 = 54% of interface bandwidth is reserved (54 Mbps) Not really interface bandwidth. When talking about buffer sizes we are talking about the sizes of the 4 queues = buffer space = memory allocation per queue. So our buffer size (which is quite small and not published but potentially 2MB per 4 ports - not important) is for Q1-4 is 10%, 10%, 26%, 54%. The bandwidth each of the 4 queues has is specified using the srr commands within the interface. In queue 2: T1 is set to 138% of bandwidth (138% x 10 Mbps) T2 is set to 138% of bandwidth (138% x 10 Mbps) T3 is always set to 100% (100% x 10 Mbps) Reserved BW = 92% x 10 Mbps Maximum Reserved BW = 400% x 10 Mbps Let's pretend our buffer per port is 1MB. Q2 has 10% of the buffer which is 100KB. However there is a twist since we are only actually reserving 92% buffers allocated to Q2. This is defined in the reserved threshold value. So really what we are reserving or guaranteeing is 92KB of buffer space for Q2. The remaining 8% goes to what is known as the common pool- which can be used by anybody (temporarily) as and when it is needed. Q2 is allowed to grab 4x the buffers if available- so the buffer size could temporarily expand to 4MB (based on our 1MB per port example). So traffic placed into Q2T1 will be dropped when Q2 is 138% full (or when Q2 has 138KB of it buffers utilized). To get to this value we would have had to borrow some of the common pool bandwidth since only 92KB is reserved. If there is no common pool bandwidth then we would have dropped traffic sooner. Same for Q2T2. Traffic place into Q2T3 will be dropped when Q2 is 400% full (or when Q2 has 4MB of its buffers utilized). To get to this value we would have had to borrow a substantial amount of common pool bandwidth. Worst case- we would drop this traffic when the reserved buffers are full (92KB). I think the Reserved and Max Reserved are what are tripping me up. My questions are: 1. If I allocated 10% using the buffers command and therefore have 10% of the interface's Reserved Memory Pool available for Queue 2, why would I then cut it down from 10% to 9.2%? 2. Does the 400 for Max Reserved mean that T1 +T2 + T3
Re: [OSL | CCIE_Voice] cBarge during SRST
I think they cannot tell you to preserve cBarge and not show ephonesMy interpretation is that if they want you to preserve this feature they will not give this restriction. -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ On Mar 5, 2012, at 6:40 PM, Ken Wyan kew...@gmail.com wrote: If they ask You are not allowed to have learned ephone details in running configuration I will configure telephony-service srst mode auto provision none srst ephone template 1 ephone template 1 softkeys remote-in-use cBarge NewCall ephone 1 privacy off ephone 2 privacy off Is there any possibility of interpretting my answer as wrong ? Ken On Tue, Mar 6, 2012 at 1:00 AM, Vik Malhi vma...@ipexpert.com wrote: You have to disable privacy at the ephone level (in IOS 12.4x). Disabling privacy at the telephony-service level and template level does not work. Therefore you must have srst mode auto-prov all in order to preserve cbarge. So I would expect that if you are required to preserve cBarge, the words do not pre-define any ephone or ephone-dn to not restrict the learned ephones/dn's from showing up in the running config. The problem I have with not creating any pre-defined ephones/dn's is that you must have a pre-defined ephone-dn for the conference ad-hoc so I'm not sure if this question is possible. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 4, 2012, at 2:50 AM, Vega Wong wrote: Hi All, Let say we need to make sure the same call feature remains during SRST, and there is a share line with cbarge during normal CUCM operation. However, if the requirement is Do not pre-define any ephone or ephone-dn in running-config how would you interupt this? The simplest way is to use srst mode auto-provision none. The issue with this command is that cBarge during SRST would not work. So if we use srst mode auto-provision all, the configure in running-config will be learnt during SRST. would that still consider as pre-define? Cheers ___ 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] BACD with the latest IOS versions
You can see which applications are running as shown below (and yes is the answer to your question). SiteC-RTR#sh call application voice summ SERVICES (standalone applications): name typedescription dsapp C Scriptbuiltin:DSESS_Service.C ipsla-responder Tcl Script builtin:app_test_rcvr_script.tcl clid_authen Tcl Script builtin:app_clid_authen_script.tcl clid_col_npw_npw Tcl Script builtin:app_clid_col_npw_npw_script.tcl AFW_THIRD_PARTY_CCC Scriptbuiltin::Third_Party_CC_Service.C CALLIndSs_SErviCe C Scriptbuiltin:CallIndSs_Service.C Default C Scriptbuiltin:Session_Service.C CTAPP C Scriptbuiltin:CallTreatment_Service.C clid_authen_col_npw Tcl Script builtin:app_clid_authen_col_npw_script.tcl fax_hop_onTcl Script builtin:app_fax_hop_on_script.tcl ipsla-testcallTcl Script builtin:app_test_place_script.tcl app-b-acd-aa Tcl Script builtin:app_b_acd_aa_script.tcl clid_authen_npw Tcl Script builtin:app_clid_authen_npw_script.tcl session Tcl Script builtin:app_session_script.tcl app-b-acd Tcl Script builtin:app_b_acd_script.tcl clid_authen_collect Tcl Script builtin:app_clid_authen_collect_script.tcl clid_col_npw_3Tcl Script builtin:app_clid_col_npw_3_script.tcl Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 2, 2012, at 1:31 PM, Rajasekar Shanmugam wrote: Experts - I would like to know , if the BACD application is supported without the TCL scripts in the flash. Meaning , is there an embedded application / script available with the later IOS releases ? -- Raj ___ 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] No Prompt for Br1 Phones
I'm sure it's no audio from anything at HQ not just CUC. If you have RSVP no sccp/sccp on both routers. Check you do not have cRTP on only one side - should be on both sides of the FR. -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 On Mar 2, 2012, at 3:38 PM, Cisco Nut rafayc...@gmail.com wrote: Hi I have integrated Unity Connections 7.x and CUCM 7.x via SCCP. When I dial VM Pilot number from HQ phone , I hear CUC Prompt, I enter my password etc, it works fine, when I dial from BR1 phones, I dont hear CUC prompt. It seems BR1 phones are able to dial VM Pilot number, I even created another DP for VM Ports and assign it to a region where it only use G711 Codec but still I am not hearing any prompts for BR1 Phones. From BR1 phones I can dial MWI number and light do go On / Off. Any Ideas!!! ___ 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] QOS and Payload Size
For (1) it doesn't make sense to me- I would expect them to state that you should use the default payload size (20ms). So I'm not sure on this For (2) with RSVP the call setup ALWAYS asks for worst case scenario which is 40kbps for g729 and after the call is answered the actual bandwidth is used (default 24kbps). So if you always used default payload size (24kbps for G729) and, say for 10 calls, use ip rsvp band 240 then this will only allow 9 calls since the last call will demand 40kbps yet there is only 24 kpbs left (and 216kpbs reserved). So you should allow one call for 10ms (even though the call never uses 10ms sampling rate) and the remainder at 20ms. So for 10 calls- (9*24) + 40. On Feb 27, 2012, at 9:05 PM, AJ BG wrote: Hello All, I have to QOS related quesitons. Scenario 1: Assuming that the lab does not instruct you to change the default payload size:however QOS requirement prompts you to calculate the priority queue bandwidth with any values other than default codec’s payload size. Then should we change the preferred packet size value in the service parameters? Or even the “Code Yellow Entry Latency” value. Will proctors expect such changes when it is not specifically mentioned? Scenario 2: if there is RSVP in the lab but the default payload size for QOS is not given to you, will you calculate a worst case scenario for one call (10ms) and the rest with default payload size? or calculate all calls with default payload size ? Thanks, AJ ___ 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] QoS 3750
When mls qos is enabed srr is enabled and cannot be disabled- since you cannot set srr share to zero. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Dec 6, 2011, at 6:19 AM, datucha123 datucha123 wrote: Hello, What is the default Queuing method on 3750, when the mls qos is enabled globally, but no srr is configured. Is it FIFO? ___ 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] cBarge in CME SRST
Make sure the shared line is an octo line. -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ On Feb 24, 2012, at 5:09 AM, Rynard Coetzee rynard.coet...@bytes.co.za wrote: I have tried all and none ,same outcome. -Original Message- From: Farkas Péter [mailto:wormh...@sch.bme.hu] Sent: 23 February 2012 03:52 PM To: Rynard Coetzee Cc: ccie_voice@onlinestudylist.com Subject: Re: RE: [OSL | CCIE_Voice] cBarge in CME SRST What srst provision in your scenario: none/dn/all? - Original Message - From: Rynard Coetzee rynard.coet...@bytes.co.za Date: Thursday, February 23, 2012 7:00 am Subject: RE: [OSL | CCIE_Voice] cBarge in CME SRST To: wormh...@sch.hu wormh...@sch.hu Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Yes I have tried it under the template and under the ephone. -Original Message- From: Farkas Péter [ Sent: 22 February 2012 02:58 PM To: Rynard Coetzee Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] cBarge in CME SRST Have you tried to turn off privacy and enable remote-in-use sofktkey through an ephone-template attached to the ephone? Privacy setting on ephone has a bug in SRST mode. Peter - Original Message - From: Rynard Coetzee rynard.coet...@bytes.co.za Date: Wednesday, February 22, 2012 1:51 pm Subject: [OSL | CCIE_Voice] cBarge in CME SRST To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Hi All I have an issue to get the cBarge to work when my H323 GW goes into SRST ,the shared line shows up on both phones ,but when I have an active call on one phone ,I don`t see the number on the other phone ,and the other phone does not go into remote in use state when I press the shared line button. I have privacy turned off under the ephones and also under the telephony service. Also my CFB is registered to the router when in srst mode ,I am able to make a normal ad-hoc conference when in srst mode. Any ideas ? Regards Rynard ___ 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] CCIE Voice Five-Lab, lab1, Q9.1
I don't think inter-digit timeout is an issue UNLESS they have mentioned that you should avoid it. In any case- I would always recommend dial-peer/voice translation instead of num-exp because you can modify Type of Number and Plan which you cannot modify with num-exp. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Feb 15, 2012, at 6:09 AM, Tomasz Pawlus wrote: If i configure: num-exp 2...$ 9001202552... in CCIE Voice Five-Lab, lab1, Q9.1 instead of a separate dial-peer pots (with prefix 9001202552) to call from SC to SA in SRST I have to wait timeout interdigit which isn't the case for a separate dial-peer. Is is possible to avoid timeout interdigit and dial immediately 2... extensions in SA without changing this parameter under telephony-service? If not, does it matter from the proctor point of view? Kind regards, Tomasz ___ 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] Query wrt to Cbarge on BR1 router without using the CME for SRST
I don't see the restriction in OWLE lab #2. The HA question 7.1 states · You should ensure that all learned ephones and ephone-dn’s appear in the running config to achieve this task. This indicates srst mode auto-provision all which is required for cBarge preservation since privacy needs to be disabled at the ephone level. Let me know the details of the lab and question number and I'll try and clear it up- but nonetheless- you are correct in what you have stated- you cannot preserve cBarge with call-manager-fallback Thanks Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Feb 15, 2012, at 4:12 PM, Rajasekar Shanmugam wrote: Experts - I`m practicing the scenario 2 from the IP Expert OWLE series the question asks us , not to use the CME based SRST for BR1. So we are forced to use the call-manager-fallback. There is a requirement later in the lab ,asking for Cbarge functionality on SRST. Wondering , if we have an option to register the hardware media resources (CFB) with the call-manager-fallback to get this working ? The solution guide suggests to configure the telephony service in order to do so. Confused here won`t that break the original requirement in the HA section , that asked us not to use the CME SRST ? Please advise. -- Raj ___ 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] Query wrt to Cbarge on BR1 router without using the CME for SRST
Bug- doesn't work- need to do it on ephone. -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 On Feb 15, 2012, at 9:30 PM, datucha123 datucha123 datucha...@gmail.com wrote: You can also configure the Privacy Settings globally, at Telephony-service configuration. with no privacy command, so that you will not need to disable it per ephone. On Wed, Feb 15, 2012 at 11:10 PM, Vik Malhi vma...@ipexpert.com wrote: I don't see the restriction in OWLE lab #2. The HA question 7.1 states · You should ensure that all learned ephones and ephone-dn’s appear in the running config to achieve this task. This indicates srst mode auto-provision all which is required for cBarge preservation since privacy needs to be disabled at the ephone level. Let me know the details of the lab and question number and I'll try and clear it up- but nonetheless- you are correct in what you have stated- you cannot preserve cBarge with call-manager-fallback Thanks Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Feb 15, 2012, at 4:12 PM, Rajasekar Shanmugam wrote: Experts - I`m practicing the scenario 2 from the IP Expert OWLE series the question asks us , not to use the CME based SRST for BR1. So we are forced to use the call-manager-fallback. There is a requirement later in the lab ,asking for Cbarge functionality on SRST. Wondering , if we have an option to register the hardware media resources (CFB) with the call-manager-fallback to get this working ? The solution guide suggests to configure the telephony service in order to do so. Confused here won`t that break the original requirement in the HA section , that asked us not to use the CME SRST ? Please advise. -- Raj ___ 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] Cbarge SRST w/ auto provision all
Are your ephone DN's octo? -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 On Feb 16, 2012, at 1:06 AM, Larry Stern larry.st...@blackbox.com wrote: In my case I could not ever get Cbarge in SRST to work, specifically Lab 1 of the new 5 lab Handbook Question 9.2. Hardware conference is registered to SC and Conf works fine. But when you hit the busy shared line and depress Cbarge, on SC PH2, you get dial tone as if a new call is to be made. I hard coded the privacy off on the ephones and even reloaded afterwards. I am not on a rack now but also tried this at work with the same result. See below. Now that I think about it, is it possible I need hunstop channel on the shared DN?? ephone 1 privacy off device-security-mode none mac-address 0026.CBBE.E8C9 ephone-template 1 button 1:3 2:1 ephone 2 privacy off device-security-mode none mac-address 0026.CBBE.EC4F ephone-template 1 button 1:4 2:1 hardware conf is registered . privacy is disabled under telephony service and in ephone. CME version 7.1 -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Wednesday, February 15, 2012 6:45 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 72, Issue 98 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: Query wrt to Cbarge on BR1 router without using the CME for SRST (Vik Malhi) 2. Connection for PSTN Hard Phone if using home Lab (Ikenna Izugbokwe) 3. Re: 2651xm instead of 2811's (Anthony Alba) 4. UCCX Session step Cisco's example script (Anthony Alba) 5. Re: 2651xm instead of 2811's (Edwin Jean-Gilles) -- Message: 1 Date: Wed, 15 Feb 2012 22:17:05 + From: Vik Malhi vma...@ipexpert.com To: datucha123 datucha123 datucha...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com, Rajasekar Shanmugam rajaseka...@gmail.com Subject: Re: [OSL | CCIE_Voice] Query wrt to Cbarge on BR1 router without using the CME for SRST Message-ID: 1721bd75-1605-4fbe-83d4-0394254d4...@ipexpert.com Content-Type: text/plain; charset=utf-8 Bug- doesn't work- need to do it on ephone. -- Vik Malhi ? CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 On Feb 15, 2012, at 9:30 PM, datucha123 datucha123 datucha...@gmail.com wrote: You can also configure the Privacy Settings globally, at Telephony-service configuration. with no privacy command, so that you will not need to disable it per ephone. On Wed, Feb 15, 2012 at 11:10 PM, Vik Malhi vma...@ipexpert.com wrote: I don't see the restriction in OWLE lab #2. The HA question 7.1 states ? You should ensure that all learned ephones and ephone-dn?s appear in the running config to achieve this task. This indicates srst mode auto-provision all which is required for cBarge preservation since privacy needs to be disabled at the ephone level. Let me know the details of the lab and question number and I'll try and clear it up- but nonetheless- you are correct in what you have stated- you cannot preserve cBarge with call-manager-fallback Thanks Vik Malhi ? CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Feb 15, 2012, at 4:12 PM, Rajasekar Shanmugam wrote: Experts - I`m practicing the scenario 2 from the IP Expert OWLE series the question asks us , not to use the CME based SRST for BR1. So we are forced to use the call-manager-fallback. There is a requirement later in the lab ,asking for Cbarge functionality on SRST. Wondering , if we have an option to register the hardware media resources (CFB) with the call-manager-fallback to get this working ? The solution guide suggests to configure the telephony service in order to do so. Confused here won`t that break
Re: [OSL | CCIE_Voice] DHCP Timeout
I would query whether the BR1 phones really do get an IP from the DHCP server- can you erase the cnf file and check from the BR1 phone (settings-*-*-#-more-erase). A good test is to create an SVI for the voice subnet and check if this gets an IP. From HQ-3750: int vlan 20 ip add dhcp Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Feb 10, 2012, at 9:47 AM, Shirley, Kris C. wrote: I know it is redundant, but if BRI phones get DHCP that is proof the Subnet for the DHCP scope in CUCM is good for that subnet only. Can you please provide the DHCP subnet screen shoot for the HQ site in CUCM? We have seen the configuration across the board for the IOS side but not CUCM. Thanks Kris From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Baktha Muralidharan Sent: Friday, February 10, 2012 10:51 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] DHCP Timeout - not sure if utils csa disable will help since it seems DHCP server IS doling out IP addresses - in fact, prbably all is ok on UCM, if it is giving out IP addresses to BR1 phones, even though they are on a different subnet - you could turn on debug ip dhcp server events on HJQ router, to see if broadcast messages requesting IP addr are going out to UCM - make sure there are no vlan restrictions on the trunk between switch and HQ router. - make sure no DHCP snooping is going on on the switch - make sure DHCP service is running on HQ router thanks, /Baktha On Fri, Feb 10, 2012 at 11:16 AM, ccie_voice-requ...@onlinestudylist.com wrote: Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: DHCP Timeout (Ramy Abdelrahim) 2. Re: DHCP Timeout (Rrcrumm) 3. Re: DHCP Timeout (Eliot Ngwa) 4. Re: DHCP Timeout (Kevin Spicer) -- Message: 1 Date: Fri, 10 Feb 2012 15:18:28 + From: Ramy Abdelrahim ramyoth...@hotmail.com To: whl...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] DHCP Timeout Message-ID: snt133-w38ba3ad777d4712b1e0046d9...@phx.gbl Content-Type: text/plain; charset=windows-1256 CUCM is pingable from both the switch and the HQ-RTR. HQ-RTR#ping 10.10.210.10 source 10.10.200.3 Type escape sequence to abort.Sending 5, 100-byte ICMP Echos to 10.10.210.10, timeout is 2 seconds:Packet sent with a source address of 10.10.200.3!Success rate is 100 percent (5/5), round-trip min/avg/max = 1/2/4 ms Date: Fri, 10 Feb 2012 08:17:42 -0600 Subject: Re: [OSL | CCIE_Voice] DHCP Timeout From: whl...@gmail.com To: ramyoth...@hotmail.com CC: ccie_voice@onlinestudylist.com OK so BR1 is working but HQ is not. Difference is that BR1 the switch ports have direct access to the routing, they are clearly being routed. My guess is that the voice vlan on HQ is not able to reach the CUCM. See if you can ping the CUCM from the switch. If this does not work, then you will have to find your routing issue. 2012/2/10 Ramy Abdelrahim ramyoth...@hotmail.com Dear All, I faced a scenario on workbook 2 that requests to have HQ and BR1 phones acquire their IP addresses from UCM-PUB. What happened was BR1 phones were able to get IP addresses from the UCM-PUB but HQ phones were not. The Switch and HQ router configuration is as follows. I appreciate if anyone can help on that. NOTE: The UCM-PUB is pingable from the switch and the HQ-RTR. Switch: vlan 10 name DATA!vlan 20 name PHONES!vlan 30 name SERVERS!interface fastethernet 1/0/1 -- To HQ router switchport trunk encapsulation dot1q switchport mode trunk switchport trunk native vlan 10!interface fastethernet 1/0/2 -- HQ Phone 1 switchport access vlan 10 switchport mode access switchport voice vlan 20 spantree portfast !/// HQ-RTR: interface fastethernet 0/0.10 encapsulation dot1q 10 native ip address 10.10.100.1 255.255.255.0!interface fastethernet 0/0.20 encapsulation dot1q 20 ip address 10.10.200.3 255.255.255.0 ip helper-address 10.10.210.10!interface fastethernet 0/0.30 encapsulation dot1q 30 ip address 10.10.210.1 255.255.255.0
Re: [OSL | CCIE_Voice] New lab 5 CUCM/CUC SIP Integration
Can you try doing a CUC/UCM SIP integration with authentication required? I'm not able to test this right now but can do later on. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Feb 10, 2012, at 9:31 AM, romain mullier wrote: Edgar, William, Vik, I recreated this issue with the CUC/SIP integration and narrowed it down a bit more. It seems that the issue only occurs when the caller does not have a mailbox on CUC (Site C callers, PSTN callers..). If you add a VM box for your Site C guy for instance then he can successfully leave the voicemail to A and B with MWI and the whole nine yard. If you remove his mailbox then you are back to the point where the message never makes to its destination. Still haven't figured out the root cause. Anyone? Romain On Sun, Feb 5, 2012 at 9:42 AM, datucha123 datucha123 datucha...@gmail.com wrote: I have the same issue in my own LAB, and as soon as I restart my CUC server, the MWI and Message start to work from PSTN for a while. but then again stops. And I make restart of CUC server every time. Thus I was using SCCP integration. On Sun, Feb 5, 2012 at 1:00 AM, Edgar Feliz ejzi...@gmail.com wrote: Also another issue I had was that it seemed like when I was leaving a VM and press # it was not recognizing that from any phone other then SA. Had most of the lab working except for the SIP/CUC. Thanks, Edgar On Sat, Feb 4, 2012 at 3:15 PM, Vik Malhi vma...@ipexpert.com wrote: Can you successfully leave SAP2 a new VM from any phone ? Another SA phone or PSTN or SB? -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ On Feb 4, 2012, at 12:08 PM, William Affeldt william.affe...@yahoo.com wrote: So I am good then? Sent from my iPhone On Feb 4, 2012, at 11:59 AM, Vik Malhi vma...@ipexpert.com wrote: Edgar- make sure that you do not have one way cRTP. Or there is any MTP being used no sccp/sccp. Bill- there is a CUC bug when you leave a VM and can press # and hear your message, but this message never gets sent to the mail box (from specific ip addresses). In this case you have to just rely on VM/MWI from an extension /gateway that does work. -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ On Feb 4, 2012, at 11:42 AM, William Affeldt william.affe...@yahoo.com wrote: You are having one way audio issues then. Sent from my iPhone On Feb 4, 2012, at 11:27 AM, Edgar Feliz ejzi...@gmail.com wrote: I don't hear the message when I press # for more options from SC or PSTN nothing is happening but I am getting the options from SA/SB E On Sat, Feb 4, 2012 at 1:44 PM, William Affeldt william.affe...@yahoo.com wrote: I am currently having the same problem. I have been troubleshooting for a hour now. It forwards to the correct VM box and you can even play the message back to your self after you record it. It just never makes it to the mailbox. From: Edgar Feliz ejzi...@gmail.com To: ccie_voice@onlinestudylist.com Sent: Saturday, February 4, 2012 10:15 AM Subject: [OSL | CCIE_Voice] New lab 5 CUCM/CUC SIP Integration I am currently working on new lab 5 and I have my CUCM-CUC integration working, for voicemail left by SA SB to SB phone and SB SA Phone I get MWI both directions. But for PSTN or SC While I can leave a VM MWI does not work and the VM does not show up when I check the inbox for either SA or SB phones for VM left from PSTN or SC. I have looked at the SIP trunk setting
Re: [OSL | CCIE_Voice] MVA problem
My guess is MOH is not working when BR1 phones is placed on hold- and this has nothing to do with Mobile Connect. Enable G729 in the IP Voice Media Streaming Application Service Parameter. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Feb 8, 2012, at 2:37 PM, Mohammed Al Baqari wrote: Hi Datucha, Now when the BR1 phone calls this HQ phone, so that the mobile phone also ring, and when the Mobile Phone picks up the call first, then the MoH works when the mobile is hunged up, on the BR1 phone. But if the HQ phone has picked up the call first and then made the Send Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and instead the ToH is heard when the mobile Phone disconnects. I suggest to set two different network moh source files. Assign one to RDP and one to HQ Phone. Then repeat your test scenarios above and lets know which MoH file is used in each case. Also, please share the traces regarding the codec part. Regards, Mohammed Al Baqari From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of datucha123 datucha123 Sent: Wednesday, February 08, 2012 11:28 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MVA problem I have the following kind of probem: I am using SLRG for Mobile Connect calls, so that that calls to users mobiles are done through local gateway (this is just for test). Now, the HQ phone has the RDP assinged with RD of his mobile phone. Now when the BR1 phone calls this HQ phone, so that the mobile phone also ring, and when the Mobile Phone picks up the call first, then the MoH works when the mobile is hunged up, on the BR1 phone. But if the HQ phone has picked up the call first and then made the Send Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and instead the ToH is heard when the mobile Phone disconnects. Also I have noticed that when the Mobile Phone picks up the call faster then the Desk Phone, the codec negotiated is g711 from BR1 phone to its local gateway through which the call went out. But if the Desk Phone at HQ picked up the call first, and then made Send calls to mobile phone, the codec stays at G729 on BR1 phone, even though the call is going out throuhg local BR1 gateway where it should use G711. ___ 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] No Audio on TEHO calls from H.323 to MGCP Gateways
Are you using MTP and/or transcoders on the gateways? Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Feb 8, 2012, at 10:38 PM, CCIEVoiceKP wrote: Yep, reloaded the GW, still no luck. It's definitely a codec issue I changed the Region seting to let HQ and BR1 communicate with g711u and viola ... works like a charm. I set it back to g729r8 between hq and br1 and no audio. My voip dial-peers to cucm bith contain voice-class codec 1 ... and that voice class has both g711u and 729r8 I'm running out of ideas . KP On Wed, Feb 8, 2012 at 9:30 PM, Ken Wyan kew...@gmail.com wrote: Did you reload gateway? On Thu, Feb 9, 2012 at 9:18 AM, CCIEVoiceKP ccievoic...@gmail.com wrote: I'm making TEHO calls from BR1 Router (H323) to HQ (MGCP). The calls connects and stays connected however there is no audio between the two endpoints. If I make the call, again TEHO, in the other direction form HQ to BR1 the call connects and there is audio. If I shut the HQ voice-port down and force the call out of the Br1 GW the call connects and there is audio. I have transcoders registered to CUCM on both gateways, they are in hte proper MRGs, MRGLs, and Device Pools. Has anyone ever run into this? I sit the lab on riday and it seems there is always something that pops up :( KP ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] MVA problem
The only difference being you have an existing call in the case of the RD and the MOH would be the second call. Can you try removing Locations CAC altogether to rule this out. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Feb 9, 2012, at 8:52 AM, datucha123 datucha123 wrote: G729 Codec is enable in IP Voice Media Streaming Application Service Parameter. And when HQ Phone calls BR1 Phone and places BR1 on hold, the MoH is played with G729 to BR1 phone without a problem. On Thu, Feb 9, 2012 at 8:31 PM, Vik Malhi vma...@ipexpert.com wrote: My guess is MOH is not working when BR1 phones is placed on hold- and this has nothing to do with Mobile Connect. Enable G729 in the IP Voice Media Streaming Application Service Parameter. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Feb 8, 2012, at 2:37 PM, Mohammed Al Baqari wrote: Hi Datucha, Now when the BR1 phone calls this HQ phone, so that the mobile phone also ring, and when the Mobile Phone picks up the call first, then the MoH works when the mobile is hunged up, on the BR1 phone. But if the HQ phone has picked up the call first and then made the Send Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and instead the ToH is heard when the mobile Phone disconnects. I suggest to set two different network moh source files. Assign one to RDP and one to HQ Phone. Then repeat your test scenarios above and lets know which MoH file is used in each case. Also, please share the traces regarding the codec part. Regards, Mohammed Al Baqari From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of datucha123 datucha123 Sent: Wednesday, February 08, 2012 11:28 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MVA problem I have the following kind of probem: I am using SLRG for Mobile Connect calls, so that that calls to users mobiles are done through local gateway (this is just for test). Now, the HQ phone has the RDP assinged with RD of his mobile phone. Now when the BR1 phone calls this HQ phone, so that the mobile phone also ring, and when the Mobile Phone picks up the call first, then the MoH works when the mobile is hunged up, on the BR1 phone. But if the HQ phone has picked up the call first and then made the Send Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and instead the ToH is heard when the mobile Phone disconnects. Also I have noticed that when the Mobile Phone picks up the call faster then the Desk Phone, the codec negotiated is g711 from BR1 phone to its local gateway through which the call went out. But if the Desk Phone at HQ picked up the call first, and then made Send calls to mobile phone, the codec stays at G729 on BR1 phone, even though the call is going out throuhg local BR1 gateway where it should use G711. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] MVA problem
Put the CAC back on and increase the ip rsvp bandwidth to a high value such as 500 (on both routers). Do a debug ip rsvp signaling on HQ and find out how much bandwidth is being requested when the problematic call is on hold. Also what Device Pool is the MOH server in? Place it inside the HQ Device Pool / HQ Region. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Feb 9, 2012, at 9:27 AM, datucha123 datucha123 wrote: Also as Vik had asked, I have removed the RSVP CAC, and the MoH begin to work and the G729 was negotiated. Well this is very strange behavior. While using CAC, BR1 is trying to negotiate G711 to MoH, and when not using RSVP CAC, G729 is negotiated. On Thu, Feb 9, 2012 at 9:16 PM, Vik Malhi vma...@ipexpert.com wrote: The only difference being you have an existing call in the case of the RD and the MOH would be the second call. Can you try removing Locations CAC altogether to rule this out. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Feb 9, 2012, at 8:52 AM, datucha123 datucha123 wrote: G729 Codec is enable in IP Voice Media Streaming Application Service Parameter. And when HQ Phone calls BR1 Phone and places BR1 on hold, the MoH is played with G729 to BR1 phone without a problem. On Thu, Feb 9, 2012 at 8:31 PM, Vik Malhi vma...@ipexpert.com wrote: My guess is MOH is not working when BR1 phones is placed on hold- and this has nothing to do with Mobile Connect. Enable G729 in the IP Voice Media Streaming Application Service Parameter. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Feb 8, 2012, at 2:37 PM, Mohammed Al Baqari wrote: Hi Datucha, Now when the BR1 phone calls this HQ phone, so that the mobile phone also ring, and when the Mobile Phone picks up the call first, then the MoH works when the mobile is hunged up, on the BR1 phone. But if the HQ phone has picked up the call first and then made the Send Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and instead the ToH is heard when the mobile Phone disconnects. I suggest to set two different network moh source files. Assign one to RDP and one to HQ Phone. Then repeat your test scenarios above and lets know which MoH file is used in each case. Also, please share the traces regarding the codec part. Regards, Mohammed Al Baqari From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of datucha123 datucha123 Sent: Wednesday, February 08, 2012 11:28 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MVA problem I have the following kind of probem: I am using SLRG for Mobile Connect calls, so that that calls to users mobiles are done through local gateway (this is just for test). Now, the HQ phone has the RDP assinged with RD of his mobile phone. Now when the BR1 phone calls this HQ phone, so that the mobile phone also ring, and when the Mobile Phone picks up the call first, then the MoH works when the mobile is hunged up, on the BR1 phone. But if the HQ phone has picked up the call first and then made the Send Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and instead the ToH is heard when the mobile Phone disconnects. Also I have noticed that when the Mobile Phone picks up the call faster then the Desk Phone, the codec negotiated is g711 from BR1 phone to its local gateway through which the call went out. But if the Desk Phone at HQ picked up the call first, and then made Send calls to mobile phone, the codec stays at G729 on BR1 phone, even though the call is going out throuhg local BR1 gateway where it should use G711. ___ 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] No Audio on TEHO calls from H.323 to MGCP Gateways
I think the reason is a codec mismatch between the transcoder and mtp. The MTP probably has g729r8 support but the transcoder lacks this codec in the list. Either add g729r8 in the transcoder on both routers or add g729ar8 within the MTP on both routers. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Feb 9, 2012, at 12:33 PM, CCIEVoiceKP wrote: Heres my xcoder / mtp layout: hq: Xcoder registered with CUCM RSVP mtp with BR2 BR1: Xcoder registered with CUCM BR2: Xcoder registered with CUCM RSVP mtp with hq I've also registered the xcoders to the gateways themselves with no luck. KP Sent from my iPhone and I have big thumbs ... So please excuse the typos. On Feb 9, 2012, at 8:34 AM, Vik Malhi vma...@ipexpert.com wrote: Are you using MTP and/or transcoders on the gateways? Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Feb 8, 2012, at 10:38 PM, CCIEVoiceKP wrote: Yep, reloaded the GW, still no luck. It's definitely a codec issue I changed the Region seting to let HQ and BR1 communicate with g711u and viola ... works like a charm. I set it back to g729r8 between hq and br1 and no audio. My voip dial-peers to cucm bith contain voice-class codec 1 ... and that voice class has both g711u and 729r8 I'm running out of ideas . KP On Wed, Feb 8, 2012 at 9:30 PM, Ken Wyan kew...@gmail.com wrote: Did you reload gateway? On Thu, Feb 9, 2012 at 9:18 AM, CCIEVoiceKP ccievoic...@gmail.com wrote: I'm making TEHO calls from BR1 Router (H323) to HQ (MGCP). The calls connects and stays connected however there is no audio between the two endpoints. If I make the call, again TEHO, in the other direction form HQ to BR1 the call connects and there is audio. If I shut the HQ voice-port down and force the call out of the Br1 GW the call connects and there is audio. I have transcoders registered to CUCM on both gateways, they are in hte proper MRGs, MRGLs, and Device Pools. Has anyone ever run into this? I sit the lab on riday and it seems there is always something that pops up :( KP ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] MVA problem
I found the problem, but another one has arise. Please inform us of the problem/solution for the benefit of others following this thread. Q1: Can you confirm that when you answer the cell phone (working scenario) which gateway is being used to get to PSTN. Q2: Can you confirm that when you answer the deskphone (not working scenario) and transfer to cell phone which gateway is being used to get to PSTN. I suspect there is different gateway being used due to you using SLRG which could explain the differences. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Feb 9, 2012, at 9:24 AM, datucha123 datucha123 wrote: I found the problem, but another one has arise. I am using RSVP as CAC between site, where the RSVP MTP has only G729 codec enabled. And also the MoH is using Unicast so that it is subject to CAC. So now when the call is picked up by Mobile Phone and then dropped, the MoH plays good, as the BR1 site is using G729 to MoH Server. But when the HQ Desk Phone picks up the call first and then redirects to Mobile phone, after hung up the MoH does not work as the BR1 phone (somehow, why it makes so I do not know) is trying to use G711 to MoH Server, where the RSVP MTP does not pass the G711 traffic and that is why the ToH is heard. Now when I have changed the BR1 to use the Local Flash MMoH, everything was working fine (MoH was heard always). But still no idea, why the BR1 is trying to negotiate G711 to MoH server after the Mobile has dropped the call (when the Desk had sent the call to Mobile). Also I can see that when the Mobile Phone picks up the call first, the BR1 Phone is using G711 (call goes through SLRG BR1 local gateway). But when the HQ Phone picks up the call first, and then sends the call to Mobile, the BR1 phone shows that it is using G729, thus the call is going though local Gateway. On Thu, Feb 9, 2012 at 2:37 AM, Mohammed Al Baqari baqari.voic...@gmail.com wrote: Hi Datucha, Now when the BR1 phone calls this HQ phone, so that the mobile phone also ring, and when the Mobile Phone picks up the call first, then the MoH works when the mobile is hunged up, on the BR1 phone. But if the HQ phone has picked up the call first and then made the Send Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and instead the ToH is heard when the mobile Phone disconnects. I suggest to set two different network moh source files. Assign one to RDP and one to HQ Phone. Then repeat your test scenarios above and lets know which MoH file is used in each case. Also, please share the traces regarding the codec part. Regards, Mohammed Al Baqari From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of datucha123 datucha123 Sent: Wednesday, February 08, 2012 11:28 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MVA problem I have the following kind of probem: I am using SLRG for Mobile Connect calls, so that that calls to users mobiles are done through local gateway (this is just for test). Now, the HQ phone has the RDP assinged with RD of his mobile phone. Now when the BR1 phone calls this HQ phone, so that the mobile phone also ring, and when the Mobile Phone picks up the call first, then the MoH works when the mobile is hunged up, on the BR1 phone. But if the HQ phone has picked up the call first and then made the Send Calls to Mobile Phone, then the MoH does not work on calling BR1 phone and instead the ToH is heard when the mobile Phone disconnects. Also I have noticed that when the Mobile Phone picks up the call faster then the Desk Phone, the codec negotiated is g711 from BR1 phone to its local gateway through which the call went out. But if the Desk Phone at HQ picked up the call first, and then made Send calls to mobile phone, the codec stays at G729 on BR1 phone, even though the call is going out throuhg local BR1 gateway where it should use G711. ___ 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] Reorder tone while + dialing from directories(missed, received)
Because you are not dialing the + number but rather the number using the access code. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Feb 6, 2012, at 11:38 AM, Mohammed Al Baqari wrote: Hi, Thanks to the team for identifying the problem so quickly. I have one confusion, recalling your test scenarios: 1) Dialing from PSTN phone to 7961 SIP phones Call works. 2) Dialing from 7961 SIP phones to PSTN numbers Call works 3) Dialing from missed/received of 7961 SCCP and other model phones Call works 4) Dialing from missed/received of 7961 SIP Doesnt work I am assuming that both tests 2/4 are matching the same route-pattern. Therefore how it was working for test 2. It shouldn't because of urgent priority. Regards, Mohammed Al Baqari -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ashwani Sent: Sunday, February 05, 2012 12:54 AM To: Vik Malhi Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Reorder tone while + dialing from directories(missed, received) Thanks Vik. Yes now I am seeing inter-digit timeout dialing from missed and received calls. Appreciate your help and pointing me to the right direction. Ashwani On 2/4/2012 3:45 PM, Vik Malhi wrote: From SIP phones calls from the directory are sent digit by digit. This is in contrast to sccp phones which send digits en bloc (as opposed to digit by digit). A route pattern such as : \+! marked as urgent priority would cause a call from the directory from a sip phone to fail. Since the plus would match and since the urgent priority has been selected the call would get sent to the gateway with just a + (which would be stripped in the case of an h323 gateway). Remove and plus route patterns marked as urgent priority- if you want to avoid inter digit timeout create route patterns without the ! and define the exact number of digits (x's). ___ 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] Standard Local RG troubleshooting: RouteGroup :RouteGroup Name= Standard Local Route Group
Things to do to when verifying a RP SLRG Place the RP in the None partition- is this a CSS/PT issue? Point the RP to a RL which does not contain SLRG but instead an actual RG such as RG-SAIf this works you have a problem with the LRG within the DPool or the device is not in the Dpool (reset device in this case too). Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Feb 6, 2012, at 9:45 PM, datucha123 datucha123 wrote: stop routing on unallocated number flag - in this particular case, this parameter has nothing to do with the actual problem. This parameter defines the rerouting option as William has already mentioned. Ricardo, try to set the Digit Analysis Complexity to Translation and Alternate Pattern Analysis. And try to look for CUCM Traces, not the DNA. On Tue, Feb 7, 2012 at 6:37 AM, William Bell w...@netcraftsmen.net wrote: Ricardo, IIRC, the stop routing on unallocated number flag was actually first introduced for ICT call flows. However, it can be applied to other call flows. In normal call handling, when the CUCM receives a notification that a call failed to complete due to unallocated number it will stop routing the call. When you flag this service param to false, CUCM will try the next trunk or gateway in the route list/route group. -Bill On Feb 6, 2012, at 8:21 PM, Ricardo Palaver wrote: Hi Emanuel !. No , it does not work .. As far as I know, this is for use AAR, or Am I wrong? Thanks ! Date: Mon, 6 Feb 2012 23:01:14 -0200 Subject: Re: [OSL | CCIE_Voice] Standard Local RG troubleshooting: RouteGroup :RouteGroup Name= Standard Local Route Group From: aedamasc...@gmail.com To: ricardo.pala...@hotmail.com CC: ccie_voice@onlinestudylist.com Hello Ricardo, You need to go to Service Parameters Call Manager, and set the option Stop Routing on Unallocated Number Flag to FALSE I hope this helps :) Emanuel Damasceno CCNP Voice On Mon, Feb 6, 2012 at 10:05 PM, Ricardo Palaver ricardo.pala...@hotmail.com wrote: Hi Folks, I am facing a problem with Standard local route group ..., it is not working and I have no idea where could I troubleshoot it. I configured as usual ... , RL - Standard Local RG In each DP, I pointed to the respective gateway (using Local Route Group param) and of course each phone with the respective device. I tried by using DNA, but it does not go further , the last point I see is RouteGroup :RouteGroup Name= Standard Local Route Group ..., As far as I know there is nothing in the service param to enable ... or there are something ? Thanks all ! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ 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] QOS LFI and BACD files
I agree with the last post. When you have used FRTS use the command show traff to verify Interval time and target rate- default to 1536 if PVC speed is not given to you. The snippet below shows what happens when FRTS is enabled- both these PVC's will need fixing. SiteA-RTR(config)#interface Serial0/0/1:0 SiteA-RTR(config-if)#frame-relay traff SiteA-RTR#sh traff Interface Se0/0/1:0.1 Access TargetByte Sustain ExcessInterval Increment Adapt VC List Rate Limit bits/int bits/int (ms) (bytes) Active 201 56000 8757000 0 125 875 - Interface Se0/0/1:0.2 Access TargetByte Sustain ExcessInterval Increment Adapt VC List Rate Limit bits/int bits/int (ms) (bytes) Active 202 56000 8757000 0 125 875 - Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Feb 7, 2012, at 7:58 AM, CCIEVoiceKP wrote: I personally would set it to a full T1 ... Bandwidth 1536 When in doubt, explain what and why to the proctor to make sure it's ok. KP Sent from my iPhone and I have big thumbs ... So please excuse the typos. On Feb 6, 2012, at 9:02 PM, AJ BG ciscoie2...@gmail.com wrote: Hello, 1. QOS question According to Vic, if you configure LFI for a subinterface in a hub and spoke environment, Your second sub interface will dopes its CIR to 56k. To solve this issue you should configure map-class for the second interface as well. I have tested this and confirmed the problem and the solution. But if the interface bandwidth is not given to you, then in what rate do you configure the second map-class? What should be your CIR and MinCIR bandwidth? 2. BACD question will it be possible that the lab requirement will be to configure BACD without giving you direct access to the BACD files? If the above scenario happen then how would you copy the files into the router. I am thinking to use CUCM. But can you even go to Cisco’s website and download BACD tar file during the exam? Any suggestion? ___ 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] OUTVIA INVIA
viazones have always been one of the most misunderstood topics- hence this email to provide some clarity. In a nutshell- Invia is always checked (for ARQ and for LRQ) before outvia but outvia is used more often. Let's look at an example. Imagine that we have UCM and CME each registered in their own independent local zone and also a third remote Backbone zone defined on another gatekeeper. Extension 4XXX is routed to the CME and international calls (numbers beginning with 011) are routed to the backbone zone. Config below: gatekeeper zone local zoneUCM abc.com zone local zoneCME abc.com zone remote BB abc.com 1.1.1.1 1719 zone prefix zoneCME 4... zone prefix BB 011* no shut Let's look at two calls. UCM GK CME and also UCM GK BB. Note - in both cases zoneUCM is the source zone and zoneCME/BB are the destination zones for the two calls respectively. If we add a CUBE to the config we can invoke the CUBE in two ways. INVIA gatekeeper zone local zoneUCM abc.com invia VIAZONE zone local zoneCME abc.com zone local VIAZONE abc.com zone remote BB abc.com 1.1.1.1 1719 zone prefix zoneCME 4... zone prefix BB 011* no shut In this instance the CUBE will be invoked for both types of calls since the source zone has been configured with an invia command. And in both types of calls that we are making the source zone is zoneUCM. Note- If we configure outvia for zoneUCM the CUBE will not be invoked since it is the invia that is used on source zones. OUTVIA With the invia configuration above we invoke CUBE for any call coming from the UCM zone. We don't care where the call is destined for- as long as the call comes from zoneUCM we invoke the CUBE. If we only wanted to invoke CUBE for calls to the backbone (and not for calls from UCM GK CME) then do as follows: gatekeeper zone local zoneUCM abc.com zone local zoneCME abc.com zone local VIAZONE abc.com zone remote BB abc.com 1.1.1.1 1719 outvia VIAZONE zone prefix zoneCME 4... zone prefix BB 011* no shut For the call UCM GK BB the CUBE is invoked since the destination zone has been configured with an outvia. For the call UCM GK CME the CUBE is not invoked since neither the source zone (zoneUCM) nor the destination zone (zoneCME) has been configured with an invia/outvia. One last thing to mention- if the source zone has been configured with an invia AND the destination zone has been configured with an outvia, the invia trumps the outvia and the outvia is not used (CUBE is not invoked twice). On Feb 7, 2012, at 12:03 PM, datucha123 datucha123 wrote: Outvia is more accurate. Invia, in most cases, is used for incoming LRQs. On Tue, Feb 7, 2012 at 11:13 PM, mercy forall mercy_for_...@hotmail.com wrote: Hi All now in outvia and invia ,, Are is it deference if i use it in local zone or remote zone ? As per Doc, outvia for any traffic leave this zone , so are this same if i use outvia in local or remote zone I need to send the call form local zone to remote zone through CUBE as local zone , what is the correct [zone remote with outvia OR with invia CUBE ] ? thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CFUR does not work
John/All, We are not running into this bug since in lab #4 we do not make the Remote Destination phone ring at all in this lab. You will hit this bug and see that CFUR will not function correctly if the RD rings (not the case in this lab). I've tested the final solution and both the requirement in 5.1 (cell phone rings into 2002 and SAP2 user sees from 3002) and 5.2 (MVA) as well as the CFUR requirement in 9.2 are all fully functional. Not to say that I don't appreciate awareness of the information you provided- thanks! Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Feb 4, 2012, at 3:16 PM, John McGaughey (jomcgaug) wrote: Hi Vik/All I’m working on Lab #4 of the new 5 labs. Quesiton 9.2. They are asking you to configure CFUR on SiteB phone 2. However this will not work because of the RDP assigned to the phone. RDP and CFUR and not supported together. See CSCtg43998. John ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] responding to emails on this list
Please don't respond to the folks trying to advertise on this list. There are certain companies that have produced a series of labs (evidently one thru six...and counting) that are trying to hijack whatever forum they can to sell their products. I think Cisco are wise to what is going on and will continue to make changes to labs to protect the integrity of the lab- this is one of the purposes of the troubleshooting aspect of the lab. We can all have the same question but the answer for each and every one of us can be different. I don't want to preach- but regardless of your opinion- if you do feel the need to response please do this unicast and not copy the list on any responses. By the way- I'm offering 45 days for the over/under for lab #7 for any takers:-) Thanks! Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.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] New lab 5 CUCM/CUC SIP Integration
Edgar- make sure that you do not have one way cRTP. Or there is any MTP being used no sccp/sccp. Bill- there is a CUC bug when you leave a VM and can press # and hear your message, but this message never gets sent to the mail box (from specific ip addresses). In this case you have to just rely on VM/MWI from an extension /gateway that does work. -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ On Feb 4, 2012, at 11:42 AM, William Affeldt william.affe...@yahoo.com wrote: You are having one way audio issues then. Sent from my iPhone On Feb 4, 2012, at 11:27 AM, Edgar Feliz ejzi...@gmail.com wrote: I don't hear the message when I press # for more options from SC or PSTN nothing is happening but I am getting the options from SA/SB E On Sat, Feb 4, 2012 at 1:44 PM, William Affeldt william.affe...@yahoo.com wrote: I am currently having the same problem. I have been troubleshooting for a hour now. It forwards to the correct VM box and you can even play the message back to your self after you record it. It just never makes it to the mailbox. From: Edgar Feliz ejzi...@gmail.com To: ccie_voice@onlinestudylist.com Sent: Saturday, February 4, 2012 10:15 AM Subject: [OSL | CCIE_Voice] New lab 5 CUCM/CUC SIP Integration I am currently working on new lab 5 and I have my CUCM-CUC integration working, for voicemail left by SA SB to SB phone and SB SA Phone I get MWI both directions. But for PSTN or SC While I can leave a VM MWI does not work and the VM does not show up when I check the inbox for either SA or SB phones for VM left from PSTN or SC. I have looked at the SIP trunk setting and do not see anything there CSS/PTs all look correct any Ideas? Thanks E ___ 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] 7940 not changing back its firmware from SIP to SCCP
I seem to remember that 7960/40 requires that you erase the configuration file (settings-3-33 I think) before you upgrade firmware. Have you tried setting your option 150 to UCM and auto registering to UCM ? -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ On Feb 4, 2012, at 10:44 AM, Emanuel Damasceno aedamasc...@gmail.com wrote: Hello Experts, I was working on a Lab the other day that had SIP and SCCP altogether. Now I am trying to use SCCP back on a few 7940s I have here, but they are not resetting. I do the normal procedure for reset to factory defaults, but the phone goes straight to SIP. It won't let me do the procedure (holding # until it blinks, 123456789*0#). After giving up on the SCCP firmware, I set it aside and played with my other phones (7975, 7965), because they were resetting just fine. Now when I try to use the SIP config again, on that specific 7940, it is not picking up its config from the SIP server (my router). You see it starts up, and on the screen it shows the old config I had in. I tried opening the configs (typing cisco as its password), erased everything, but still no luck. Anybody has ever seen this? Anybody could give me an idea of how to fix this? Thanks. Emanuel Damasceno CCNP Voice ___ 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] New lab 5 CUCM/CUC SIP Integration
Can you successfully leave SAP2 a new VM from any phone ? Another SA phone or PSTN or SB? -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ On Feb 4, 2012, at 12:08 PM, William Affeldt william.affe...@yahoo.com wrote: So I am good then? Sent from my iPhone On Feb 4, 2012, at 11:59 AM, Vik Malhi vma...@ipexpert.com wrote: Edgar- make sure that you do not have one way cRTP. Or there is any MTP being used no sccp/sccp. Bill- there is a CUC bug when you leave a VM and can press # and hear your message, but this message never gets sent to the mail box (from specific ip addresses). In this case you have to just rely on VM/MWI from an extension /gateway that does work. -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ On Feb 4, 2012, at 11:42 AM, William Affeldt william.affe...@yahoo.com wrote: You are having one way audio issues then. Sent from my iPhone On Feb 4, 2012, at 11:27 AM, Edgar Feliz ejzi...@gmail.com wrote: I don't hear the message when I press # for more options from SC or PSTN nothing is happening but I am getting the options from SA/SB E On Sat, Feb 4, 2012 at 1:44 PM, William Affeldt william.affe...@yahoo.com wrote: I am currently having the same problem. I have been troubleshooting for a hour now. It forwards to the correct VM box and you can even play the message back to your self after you record it. It just never makes it to the mailbox. From: Edgar Feliz ejzi...@gmail.com To: ccie_voice@onlinestudylist.com Sent: Saturday, February 4, 2012 10:15 AM Subject: [OSL | CCIE_Voice] New lab 5 CUCM/CUC SIP Integration I am currently working on new lab 5 and I have my CUCM-CUC integration working, for voicemail left by SA SB to SB phone and SB SA Phone I get MWI both directions. But for PSTN or SC While I can leave a VM MWI does not work and the VM does not show up when I check the inbox for either SA or SB phones for VM left from PSTN or SC. I have looked at the SIP trunk setting and do not see anything there CSS/PTs all look correct any Ideas? Thanks E ___ 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] Reorder tone while + dialing from directories(missed, received)
From SIP phones calls from the directory are sent digit by digit. This is in contrast to sccp phones which send digits en bloc (as opposed to digit by digit). A route pattern such as : \+! marked as urgent priority would cause a call from the directory from a sip phone to fail. Since the plus would match and since the urgent priority has been selected the call would get sent to the gateway with just a + (which would be stripped in the case of an h323 gateway). Remove and plus route patterns marked as urgent priority- if you want to avoid inter digit timeout create route patterns without the ! and define the exact number of digits (x's). -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ On Feb 4, 2012, at 12:27 PM, Ashwani ash_r...@hotmail.com wrote: Hello Everyone, I am having issue + dialing (missed, received) from HQ, BR1 ( 7961 SIP ) phones. I can dial from 7961 SCCP phones , other 7965 and 7962 phones. Can someone please explain me why I am not able to dial from SIP phones? Here are the testing I have done so far.. 1) Dialing from PSTN phone to 7961 SIP phones Call works. 2) Dialing from 7961 SIP phones to PSTN numbers Call works 3) Dialing from missed/received of 7961 SCCP and other model phones Call works 4) Dialing from missed/received of 7961 SIP Doesnt work Any help will be appreciated. Thanks, Ashwani ___ 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] SLRG
You are not missing anything with this unusual requirement. The real power of SLRG is scalability - and scalability is something they cannot test with 3 sites so it's just a test of how well you know the technology. On Feb 1, 2012, at 11:20 AM, CCIEVoiceKP wrote: Assume the requirements state that : a. Site1 local calls should use Site1 Router and send 7digits to PSTN b. If Site1 Router is Unavailable use Site2 Router and send 10 digits as a LD call c. Use local route groups for Site 1 Does this even make any sense? I suppose I could put the SLRG as the first RG in the above Route list followed of course by the RG for Site2 Router. But in my mind this really doesn’t show the power or purpose of an SLRG. Or am I just totally missing something here? Thoughts? KP ___ 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] mlpp vs frf.12
I can't see this question in the lab you mentioned. Either way FRF.12 is more efficient since you don't have any additional PPP overhead. MLP LFI would be used when you have FR on the spoke and ATM at the hub- FRF.12 is not possible in this instance. In our case both ends of the pipe are FR and therefore FRF.12 is possible and is more efficient. On Feb 3, 2012, at 5:51 AM, Farkas Péter wrote: Gents, Qos in wb2/6.2 requires the most efficient lfi technique. SG selected MLPP. Why? What is the main advantage one to the other? Thanks, Peter ___ 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] load command
Correct. In addition you should be careful NOT to erase factory defaults on the CME phone too- this would mean you need the load command in order for it to boot up (phone would need to TFTP a .loads file. On Feb 3, 2012, at 9:22 AM, Ken Wyan wrote: For CCME , it's required to use load command as below. tftp-server flash:PHONES/SCCP.loads alias SCCP.loads telephony-service load 7965 SCCPx But in CCIE lab environment , this may consume lot of time for phone firmware upgrades. I think it's better not to put any of above commands in CCIE lab unless any problem arises. (In Lab all phones should be already having v7 compatible phone loads) Do you agree with me or not? ___ 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] 5-lab Workbook #4 - Unity Connection Notification
They have been uploaded. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Jan 24, 2012, at 11:08 AM, Edgar Feliz wrote: Vic, have the updated guides been uploaded? We only get so many print cycles and just want to make sure I am not wasting paper too. Thanks On Tue, Jan 24, 2012 at 2:38 AM, Vik Malhi vma...@ipexpert.com wrote: I don't think they are in beta mode. I accidentally skipped a question, it's been corrected and it should have been uploaded. I'll ensure it is tomorrow. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Jan 22, 2012, at 7:14 PM, Jurassic Labs wrote: ! ALL BE ADVISED The new 5-lab handbooks are somewhat in beta mode I believe as I've ran into some minor errors and such. But the biggest issue is that I'm finding the solution guide for Lab#5 is not available (the estimated timeline for it to published has passed) and I'm finding out in Lab#4 there a section completly skipped in the solution guide. If anyone has some thoughts about making this work - please post. Ensure that everytime somebody leaves user [fill in name] a new voicemail in his account that a message notification call is made out to PSTN line 2. The call should be sent out of the SiteA gateway as a local call, with the SiteB gateway acting as a backup. If the call goes out the SiteB gateway, it should be a long distance call. Ensure that Unity Connection cannot place a call outbound to any other number except PSTN line 2 (+12024678124). Setting up a subscriber's notification device is the easy part - getting Unity Connection to dial out that number to CUCM - where / how is that controlled? ___ 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] 5-lab Workbook #4 - Unity Connection Notification
I don't think they are in beta mode. I accidentally skipped a question, it's been corrected and it should have been uploaded. I'll ensure it is tomorrow. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Jan 22, 2012, at 7:14 PM, Jurassic Labs wrote: ! ALL BE ADVISED The new 5-lab handbooks are somewhat in beta mode I believe as I've ran into some minor errors and such. But the biggest issue is that I'm finding the solution guide for Lab#5 is not available (the estimated timeline for it to published has passed) and I'm finding out in Lab#4 there a section completly skipped in the solution guide. If anyone has some thoughts about making this work - please post. Ensure that everytime somebody leaves user [fill in name] a new voicemail in his account that a message notification call is made out to PSTN line 2. The call should be sent out of the SiteA gateway as a local call, with the SiteB gateway acting as a backup. If the call goes out the SiteB gateway, it should be a long distance call. Ensure that Unity Connection cannot place a call outbound to any other number except PSTN line 2 (+12024678124). Setting up a subscriber's notification device is the easy part - getting Unity Connection to dial out that number to CUCM - where / how is that controlled? ___ 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] 5-lab SelfStudy NTP
All references to the backup NTP server have been removed. There were some problems with ntp version mismatches and detection of a falseticker. Please download all files related to the 5 lab handbook in 24 hours- it seems like some files have not been updated and I'll ensure they are tomorrow. So download all files related to the handbook include the labs, solutions, scripts and startup configs. I've tried to correct as many errors as possible- thanks to the people who reported problems. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Jan 23, 2012, at 12:58 PM, Jurassic Labs wrote: In both LAB#4 and LAB#5 of the newer 5-lab series, there's an NTP section that I don't think is being fully addressed in the DSG (Solution Guides). Here's the question: Configure the UCM, Site A / B / C gateway to synchronize their clock with the clock source on the backbone NTP server. The IP Address of the NTP server is 10.10.100.2. In all cases the backup NTP server is SiteA gateway loopback IP Address (10.10.110.1). The DSG just covers pointing the PUB, SiteA-RTR, SiteB-RTR, and SiteC-RTR to 10.10.100.2 via command ntp server 10.10.100.2. But isn't that just half of the story?? Shouldn't the PUB have both 10.10.100.2 AND 10.10.110.1 listed? Then SiteA would have ntp master 6, then for both SiteB SiteC we could enter the two commands, ntp server 10.10.100.2 prefer ntp server 10.10.110.1. Wouldn't that be the complete solution? ___ 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] Intersite calls MOH
We set the VGW as a multicast server- so this works for IP phones and PSTN calls. You must ensure you specify the Voice SVI IP Address and Loopback address in the multicast moh route VlanXX LoXX statement within telephony-service/call-manager-fallback. Technically this is not multicast routing- there is no routing since the VGW is itself the MOH server. You can disable multicast-routing / PIM and it will still work. On Jan 18, 2012, at 8:34 AM, brajesh kumaR wrote: Hello, If branch site VGW configured for MOH multicast from branch router/VGW so will inter site IP calls ( two different location calls) will also get multicast from voice gateway or moh from multicast only works for external PSTN calls. Will on-net calls use VGW multicast MOH from flash?? During inter site calls I found following on VGW. Multicast IP showinng 0. Jan 18 14:40:52.381: moh_update_rtp: callID 55764 dstCallID 55765 Jan 18 14:40:52.381: moh_process_ccb: dstadr 0.0.0.0, callid 55765, port 0, codec 65535, moh_en 0, moh_addr 0.0.0.0 Jan 18 14:40:52.381: moh_update_rtp: callID 55764 dstCallID 55765 Jan 18 14:40:52.473: moh_update_rtp: callID 55764 dstCallID 55765 Jan 18 14:40:52.497: moh_process_ccb: dstadr 10.136.170.197, callid 55765, port 20650, User phone IP address : codec 5, moh_en 0, moh_addr 0.0.0.0 .Multicast address coming as 0. Jan 18 14:40:52.497: moh_update_rtp: callID 55764 dstCallID 55765 Jan 18 14:40:52: %ISDN-6-CONNECT: Interface Serial0/0/0:27 is now connected to 0170480054 N/A Jan 18 14:40:52.533: moh_update_rtp: callID 55764 dstCallID 55765 Jan 18 14:40:52.537: moh_process_ccb: dstadr 10.136.170.197, callid 55765, port 20650, codec 5, moh_en 0, moh_addr 0.0.0.0 Jan 18 14:40:52.537: moh_update_rtp: callID 55764 dstCallID 55765 ___ 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] IOS version of Proctorlabs Routers
HQ is running 12.4(15)T due to gatekeeper licensing restrictions. So you have to use sccp version 5. -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 On Dec 25, 2011, at 10:47 AM, Ken Wyan kew...@gmail.com wrote: Dear All, In proctorlabs CCIE Voice Racks , they claim All routers run 12.4.22T IOS . https://proctorlabs.com/index.cfm/product/sku/CCIE_Voice_vRack_Online_Hardware_Rental_Session Actually they use older IOS images ( HQ , BR1 BR2 routers have different IOS versions ) . ( I can't copy sccp ccm 10.10.210.11 identifier 1 version 7 command between Routers. Routers expect version 5 , 6 7 depending on IOS) Why do they show false information on IOS before scheduling / purchasing rack rental sessions. ( I apologize if this is not the correct place for such complains ; but I'm sure relevant people will get my message from this list ) Thank You ___ 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] How to verify dtmf in voice gateway
On an H323 GW use debug h245 asn1 H.245 ASN1 Messages debugging is on SiteB-RTR# .Dec 17 02:38:53.994: H245 MSC OUTGOING PDU ::= value MultimediaSystemControlMessage ::= indication : userInput : alphanumeric : 4 DTMF 4 presses When using rtp-nte: .Dec 17 02:38:53.994: H245 MSC OUTGOING ENCODE BUFFER::= 6D400134 .Dec 17 02:38:53.998: On an MGCP gw use debug mgcp packet Dec 16 23:37:22.985: MGCP Packet sent to 10.10.210.11:2427--- NTFY 868796016 S0/SU0/DS1-0/1@SiteA-RTR MGCP 0.1 N: ca@10.10.210.11:2427 X: 1 O: D/1 DTMF 1 pressed --- When RTP-NTE is being used: deb voip rtp session named-event (no example). Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Dec 14, 2011, at 8:29 AM, brajesh kumaR wrote: Use debug voip ccapi inout and look for digit= to verify DTMF digits sent. You can debug this live on gateway and verify any DTMF digits entered in between. On Fri, Dec 9, 2011 at 12:38 PM, So Gwaai sogw...@gmail.com wrote: Anyone know how to verify the voice gateway send the dtmf through the PRI port? Any debug command or ccm trace we can get? ___ 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] Five-Lab Lab3 Task 4.5: GK to remote GK succeeds (supposed to fail)
The PSTN should have the command no gateway configured. Does your base config not include this command? Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Dec 13, 2011, at 7:09 AM, Anthony Alba wrote: Hi, in this task, the call from SA GK to remote PSTN GK is supposed to fail (dialing 0119167) and we are supposed to use asn1 debugs to see the LRJ (no route to destination). BUT.. My call actually succeeded. My question: is the un-bug in the initial PSTN config that is too liberal? Should there be lrq reject-unknown-prefix in the initial configuration to achieve the aim of the task? gatekeeper zone local backbone ipexpert.com 10.10.100.2 zone remote US ipexpert.com 10.10.110.1 1719 zone prefix backbone 44* gw-type-prefix 1#* default-technology no shutdown !--- call actually succeeds; 01191* is routed to local zone ! Anthony ___ 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] Five-Lab Lab 2: To +442077966596 on Phone Display - utter weirdness
I'm familiar with this- I don't know if it is by design like the affects of using Called Transformations at both the Route Pattern and Route List. It's good to know about, I think it's just a lot easier to do it without gateway called party transformation patterns (kind of defeats the object of Called Party Transformation Patterns when you have to perform manipulations on the RP or RL in combination with gw called party transformations). Vik On Dec 12, 2011, at 6:04 AM, Anthony Alba wrote: Hi, This is Lab 2 in the Five-Lab Handbook. The the task: on SC (UK site +442077964XXX, MGCP gateway) the requirement is to plus dial from directory without EditDial +442077966596 but the phone display must show To +442077966596 Normally globalized dial plan will not work; if I have one \+.! route pattern and a gateway Called Party Transformation \+4420.! -- DDI (send 8D out to PSTN) then the caller will see To 77966596. To satisfy this type of task we should use Route Pattern and digit manipulation at Route List Details. E.g. Route Pattern: \+442077966596 The route list for this task has RG_SC has primary and RG_SB as backup SB is an H.323 gateway Route List Details: RG_SC use Mask RG_SB use Mask 90114420 Caller sees To: +442077966596 But during my testing I came across a strange result where I used globalized dialplan/gateway called party transformations but got the correct display !!?? I expected it to FAIL and show To 77966596 The weirdness: if you use both global dial-plan/Called Party Transformation and at the same time use Route Pattern / Route List Details; provided the manipulation at RL details and Called Party Transformation give *identical* results then the phone will show the number as at the Route Pattern stage. Is this a bug or feature?? Example: A. WRONG: Configure only globalized dial plan +442077966596 --- 7796596: See To: 77966596 B. CORRECT: Configure both globalized dialplan and an identical overlapping route-pattern/RL details see To: +442077966596 C. TESTING: We know that gateway Called Party Transformation trumps; so to test configure globalized dialplan (correct DNIS) and deliberately create a bad route-pattern/RL details Global Dialplan +442077966596 --- 77966596 Erroneous RL details: +442077966596 --- Since Called Party Transformations trumps, we get DNIS correct and the display shows To 77966596 Summary: gateway Called Party Transformation always trumps so we always get a 8 Digit DNIS; but if Route List Details digit manipulation gives the identical pattern to the Called Party Transformation then the caller's phone will see the DNIS at the Route Pattern stage. Have you folks ever heard of this behaviour?? Anthony ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Lab 7A Workbook 1
You can configure the site with a Dpool with a UCM Group containing only the SUB. The stop the SUB UCM service. -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 On Dec 3, 2011, at 10:01 AM, Emanuel Damasceno aedamasc...@gmail.com wrote: Hey guys, I am wondering now how I should test SRST if I am not using any phones from Proctorlabs. I am using all hardware phones on my side. Any ideas? Emanuel Damasceno ___ 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] Strange behavior on MGCP gateways
Try the command no mgcp timer receive-rtcp (then no mgcp/mgcp) to see if this fixes it. On Dec 2, 2011, at 4:19 PM, Matthew Saskin wrote: I'd look a bit further, the silence is not causing the call to terminate - what about calls being muted, etc. What do the q931 debugs show when the call gets disconnected? On Thu, Dec 1, 2011 at 11:02 PM, ccielabrat ccielab...@gmail.com wrote: I noticed a weird thing while testing MGCP. If I call out to my pstn phone and answer the call by pressing the answer soft key , the call will disconnect after about two minutes. If I answer the call with the speaker button , it stays up forever. I'm guessing the problem is I don't have handsets on any of my phones , so when I answer with the answer softkey , the phone is off hook and sending dead air packets. Somehow , mgcp see this as an error condition after two minutes or so and kills the call. If the call is on speaker, I guess it picks up enough noise to keep the call from being considered inactive. Weird. ___ 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] WAN QOS Strategy Question.
This is correct. It's not AutoQoS that causes the problem- its because you MUST have FRTS enabled in order for the map-class to be attached to the DLCI. And this must happen since the service policy is inside the map-class. I recommend you run AutoQoS on all routers when doing WAN QoS or at the very least attach a map-class to all DLCI's. Also be careful if you are using cRTP. You should ensure that if you are using cRTP that both ends of the pipe are configured with cRTP otherwise you will experience one way audio. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Dec 3, 2011, at 10:59 AM, ccielabrat wrote: Hey everyone, I can confirm after A LOT of testing, if you are given a QOS requirement for only one of the two Frame PVCs, and you use Auto QOS, you will have a problem. Auto QOS will automatically config Frame-relay Traffic shaping on the physical WAN interface and then configure the PVC you are QOS'ing to the bandwidth that is noted under the sub interface. The other sub interface gets left with the default frame-relay traffic shaping behavior which is to drop CIR to 56k on the PVC. Do a Show Frame PVC dlci# on both PVC's after running auto qos. I think this could be an intended Rat hole on the exam. If you only have to configure Hq-SB QOS and you don't know much more the to run autoqos and tweak a couple parameters, your SC communications will start to fail with a PVC CIR of 56k. On Sat, Dec 3, 2011 at 10:00 AM, Ken Wyan kew...@gmail.com wrote: Hi, I think you got 56k value from this document which was published in 2005 with IOS version 11. (somehow same age as QoS SRND) http://www.cisco.com/en/US/tech/tk713/tk237/technologies_configuration_example09186a00800942f8.shtml I think it's better to put auto qos voip or auto qos voip fr-atm in the remaining interface as well (without any bandwidth as it's not mentioned in exam). Then itll take 1.5M by default. Is there a command to verify that FRTS use 56k bandwidth because above documents are very old. Ken On Sat, Dec 3, 2011 at 8:54 AM, ccielabrat ccielab...@gmail.com wrote: To All, I've been trying to figure the best/fastest way to get a WAN QOS requirement completed on exam day. I've become very comfortable with Auto-QOS and making the needed tweaks, so Auto-QOS is the way I'm going to use. The one piece of the strategy that I'm stilll wondering about is if WAN QOS is specified for only one of the PVC's. Auto-QOS will automatically put Frame-relay traffic shaping on the physical interface which has the side effect of leaving the other pvc with a 56k PVC speed. My solution here is to create a frame-relay map-class with the following parameters. map-class frame-relay Not56k frame-relay traffice-rate 1536 I apply this map-class to the other sub-interface/PVC which negates the 56k problem. I'm curious if anyone has an opinion on the Pros/Cons of this approach and if it might negate requirements somehow. ___ 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] Vol 2 Lab 7 TEHO - phone display issue
Just a clarification here. voice service voip no supplementary-service h225-notify cid-update command reference implies this is for CallerID updates when in fact it is for Called # updates The above prevents an H323 GW from updating the Called Number on the Callers phone. So with the following example we can ensure the caller always sees TO 4158884343. RP: 91415.8884343 - called # mask = 10X's This is for the display on the callers phone for when the call leaves the BR1 or HQ gateway. RL: br1h323-h1mgcp RG: BR1 - called #: Predot, Prefix 9 This is to match the local dial-peer in BR1 RG HQ: called #: Predot, Prefix 1415 This is here to satisfy the HQ Telco requirements. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Dec 3, 2011, at 8:14 AM, John McGaughey (jomcgaug) wrote: Hi Stuart, I ran into the exact same problem. The answer in the DSG is incorrect. I got it to work by doing the following. I created the following dial-peer on BR1. dial-peer voice 415 pots destination-pattern 415... port 1/0/0:23 forward-digits 7 In the teho route list I set the called party transform mask to XX for the BR1 RG. Now I see TO: 4158884343 on the display when the call does to the BR1 GW. “no supplementary-service h225-notify cid-update” is not needed since that is for calling party, not called party. I did leave XX in the called party transformation mask in the route pattern because that is needed for calls going out the HQ MGCP GW. HTH John From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Geoghegan, Stuart Sent: Friday, September 23, 2011 7:44 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Vol 2 Lab 7 TEHO - phone display issue Hi All, I am working through Vol2 Lab TEHO requirement where we are asked to dial 914158884343 from HQ Phone via the BR1 gateway (H323) as first choice as a local call. and then the HQ MGCP Gateway as second choice as National call 1 + 10 digits. I have no issue with the routing of the calls and digit manipulation, however the question requests “the caller should see TO 4158884343 regardless of whether the call is sent out of Br1 HQ. I have read the PG and OSL Archives. I have tried using called party transformations XX on the Route Pattern level also the RG level. I am definitely sending 4158884343 to both gateways, however I my display always states “TO 914158884343”. Previous OSL archives suggest adding the following to the H323 GW voice service voip no supplementary-service h225-notify cid-update I have tried this but this has not changed anything – and also I still have the same issue via my MGCP gateway. I have even tried a TP of 914158884343, called party mask of 4158884343, which then matches a RP of 4158884343 – which again routes correctly but the display does not change. Has anyone else encountered this before? kind regards Stuart The information contained in this e-mail (and attachments) is confidential. It must not be read, copied, disclosed, printed, forwarded, relied upon or used by any person other than the intended recipient. Unauthorised use, disclosure or copying is strictly prohibited. If you have received this e-mail in error immediately contact the sender andpostmas...@ngbailey.co.uk then permanently delete it and any attachments. NG Bailey may monitor email where it is considered proportionate to any perceived risk. This email has been sent on behalf of an NG Bailey company, if in doubt ask the sender to clarify which company. The NG Bailey companies include NG Bailey Limited (342778), NG Bailey IT Services Limited (2338401), NG Bailey Facilities Services Limited (5472032), Kedington (Northern Ireland) Limited (NI 31145) and NG Bailey Group Limited (1490238). All of the companies, except Kedington (Northern Ireland) Limited, are registered in England with registered office at Denton Hall Ilkley West Yorkshire LS29 0HH. Kedington (Northern Ireland) Limited is registered in Northern Ireland and its registered office is c/o Carson McDowell, Murray House, Murray Street, Belfast, BT1 6DN. ___ 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] 5 New labs what to watch for!!!
The whole point of the question is to get you to dial the wrong number and explain the proem. So what is in the lab question is correct. -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 On Nov 27, 2011, at 11:31 AM, Edgar Feliz ejzi...@gmail.com wrote: Sorry had a copy/paste issue myself 2nd number in first point should be 011916745738932 On Sun, Nov 27, 2011 at 12:12 PM, Edgar Feliz ejzi...@gmail.com wrote: Guys I have been working on the new (5) labs mostly 1-3 since they have the guide. Here is what i have found to be things to watch for and fixes, Please note i have been working on these for over 10 days and it's the same every time it is just the way the integration was done for the VM images purposely or not. There are some typos you will see reference to 011916745738931 but if you try to call that number it is not available true number is 011916745738931 There are some others name of routers in guide not always right when you look for solutions so just make sure you note what router you are working on. I think this is mainly because of copy and pasting from one lab to another etc since some things are similar but devices are different. UCCX integration; rmjtapi app-user is not configured add rmjtapi add phones and cti enable make sure pwds match on CCX and CUCM for jtapi_1 and rmjtapi from CCX run sync and restart engine You may have to re-sync a couple of times the T flag is not set so the CUCM never sees that the CCX is integrated you have to do a manual step to get this up and running so you can add the UCCX extension to users, run this command on the CUCM Pub as stated in earleir emails run sql update processconfig set paramvalue=T where paramname like '%nstalled%' NO need to reload it takes affect right away CUC - I have had to reload the CUC server every time for MWIs and to be able to leave message... This one I am not sure why I have no fix other than to restart the server. CUE - As noted in labs WB and Guide you will need to change licenses depending on the lab you are working on. Lab 2 CCXquestion there is a step missing in the script. a generated prompt of zero calls ahead of you when you are the only one in queue = you do need the decrement step (thanks larry) If i recall or run into any more items of interest i will post back i am in last 5 days of prep and will be concentrating on the labs E ___ 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] 5 New labs what to watch for!!!
Ok - thanks- will fix this. -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 On Nov 27, 2011, at 11:41 AM, Larry Stern larry.st...@blackbox.com wrote: I also found in my rack 33, the pstn line 3 for calling number only sends 408 I had to modify this in the PSTN router rule 2 from /^\(408\)8397263$ to /^408\(8397263)/ /\1/. Try typing that fast on an IPAD(lol) This was for lab 1and 2. Sent from my iPad On Nov 27, 2011, at 12:12 PM, Edgar Feliz ejzi...@gmail.com wrote: Guys I have been working on the new (5) labs mostly 1-3 since they have the guide. Here is what i have found to be things to watch for and fixes, Please note i have been working on these for over 10 days and it's the same every time it is just the way the integration was done for the VM images purposely or not. There are some typos you will see reference to 011916745738931 but if you try to call that number it is not available true number is 011916745738931 There are some others name of routers in guide not always right when you look for solutions so just make sure you note what router you are working on. I think this is mainly because of copy and pasting from one lab to another etc since some things are similar but devices are different. UCCX integration; rmjtapi app-user is not configured add rmjtapi add phones and cti enable make sure pwds match on CCX and CUCM for jtapi_1 and rmjtapi from CCX run sync and restart engine You may have to re-sync a couple of times the T flag is not set so the CUCM never sees that the CCX is integrated you have to do a manual step to get this up and running so you can add the UCCX extension to users, run this command on the CUCM Pub as stated in earleir emails run sql update processconfig set paramvalue=T where paramname like '%nstalled%' NO need to reload it takes affect right away CUC - I have had to reload the CUC server every time for MWIs and to be able to leave message... This one I am not sure why I have no fix other than to restart the server. CUE - As noted in labs WB and Guide you will need to change licenses depending on the lab you are working on. Lab 2 CCXquestion there is a step missing in the script. a generated prompt of zero calls ahead of you when you are the only one in queue = you do need the decrement step (thanks larry) If i recall or run into any more items of interest i will post back i am in last 5 days of prep and will be concentrating on the labs E This email and any files transmitted with it are confidential and are intended for the sole use of the individual to whom they are addressed. Black Box Corporation reserves the right to scan all e-mail traffic for restricted content and to monitor all e-mail in general. If you are not the intended recipient or you have received this email in error, any use, dissemination or forwarding of this email is strictly prohibited. If you have received this email in error, please notify the sender by replying to this email. ___ 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] AAR and Unity Connection
The AAR GRP can be empty if you have the full e164 number in the external number mask of your internal DN's. You can then create Route Patterns beginning with the + sign. If you had say 10 digits External Number Masks on your DNs then you could use the AAR GRP prefix to insert an access code to match on Route Patterns- eg prefix 9 or 91. -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ On Oct 29, 2011, at 10:42 PM, Randall Crumm rrcr...@yahoo.com wrote: Hi, I am adding the AAR group in CUCM. The name is CUCM. Do I need to put anything in the prefix digits within CUCM? Is there a rule as to when you would add something? Why/when would you add something? 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
Re: [OSL | CCIE_Voice] CUE-2-CCM Can't dial VM 3600
Well if the CTI Ports are down that is why it is not working. You HAVE to do the UCM work before the CUE side of the integration. You should add the CTI Ports- and supply the jtapi user credentials. This jtapi user should be associated to the cti rp and cti ports and have standard cti enabled. ccn subsystem jtapi ctiport 3601 3602 ccm-manager address 10.10.210.11 10.10.210.10 ccm-manager credentials .. end subsystem Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Oct 28, 2011, at 7:31 PM, edgar feliz wrote: Hi Vic, thanks for the response, here is the some of the output fomr the show command, The VM (3600) port shows registered the other two (3601/3602) do not. Also all I get is a fast busy when I try to dial the 3600 or press the vm service button. ccn trigger jtapi phonenumber 3600 application voicemail enabled locale en_US maxsessions 6 end trigger ccn subsystem jtapi ccm-manager address 10.10.210.11 10.10.210.10 ccm-manager credentials hidden OSGGx1TGBhk3jDxWVhfWhUnfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmP end subsystem From: Vik Malhi vma...@ipexpert.com To: edgar feliz ejfeli...@yahoo.com Cc: CCIE ccie_voice@onlinestudylist.com Sent: Friday, October 28, 2011 10:02 PM Subject: Re: [OSL | CCIE_Voice] CUE-2-CCM Can't dial VM 3600 Are CTI Rps and CTI ports registered? When you sh run inside the cue cli do you see the CTI ports under ccn subsystem jtapi? Also check the ccn trigger jtapi phonenumber Do you hear annunciatior or fast busy? More info pls -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 On Oct 28, 2011, at 5:29 PM, edgar feliz ejfeli...@yahoo.com wrote: I have set up CUE to CCM configured CTI ports, vm profile etc. double checked with all the IE guide and can't get BR2 phone to dial into VM? Any Ideas? Users imported int0 CUE OK, CCM can see the 10.10.202.2 CUE when I look at CTI RPs. EJF ___ 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] Vol 1 LAB 5.2 -- SoS
Clearly this is a problem with SLRG. When you set the LRG on the DP reset the DP. Also reset the RL that contains SLRG... On Oct 28, 2011, at 5:25 AM, Voiper wrote: Hi Rrcrumm: Appreciate your suggestions and thankful. For my IB dial-peer, here is my config dial-peer voice 5000 voip destination-pattern 2123945... (also tried with 5...) voice-class codec 1 session target ipv4:10.10.210.11 incoming called-number . (also tried separate dial-p 5002 with just incoming called-number .) dtmf-relay h245-alphanumeric no vad When I dail 911 to PSTN I get the busy signal tone. If I don't use the Standard Local Route Group and simply point the RP 911 to 10.10.200.3 or BR1-RTR.ipexpert.com, then the calls to 911 from hq and br1 phones are successful. The Lab 5A Final config in IPX shows these commands: voice service voip sip bind control source-interface FastEthernet0/0.20 bind media source-interface FastEthernet0/0.20 I am not sure if I have them. I will check them in the evening after work. Thanks and have a good one Voiper On Fri, Oct 28, 2011 at 2:39 AM, Rrcrumm rrcr...@yahoo.com wrote: Ahhh How about your IB dial peer. Try 5... As a destination pattern or just create a new one with 5... Randall Sent from my iPhone On Oct 27, 2011, at 4:14 PM, Voiper datapack...@gmail.com wrote: Any help most welcome Voiper On Thu, Oct 27, 2011 at 6:44 PM, Voiper datapack...@gmail.com wrote: Thanks John for the prompt reply and suggestion. * rg-hq was created for HQ rg-rl was created for BR1 * rl-local-gw standard local route group * RP 911 rl-local-gw * DP hq rg-hq br1 rg-br1 * inbound dial-p voice 100 with incoming number . configured dial-peer voice 1 pots incoming called-number . direct-inward-dial dial-peer voice 5000 voip destination-pattern 2123945... voice-class codec 1 session target ipv4:10.10.210.11 incoming called-number . dtmf-relay h245-alphanumeric no vad dial-peer voice 911 pots destination-pattern 911 port 0/0/0:23 forward-digits 3 * Serial0/0/0:23 unassigned YES NVRAM up up * HQ-RTR#sh contro t1 T1 0/0/0 is up. Applique type is Channelized T1 * Call Manager service rebooted (last effort) * debug isdn q931 and debug voip dial-peer shows nothing dial 911 from HQ-ph2 and BR1-ph2 nothing happening. I just don't know what am I missing? It is a pretty straight forward lab, infact the beginning of Call Routing! ! Voiper On Thu, Oct 27, 2011 at 2:29 PM, John Ciccone ccie.cicc...@gmail.com wrote: Voiper, Go back and verify your steps 1 and 2. 1) Created the RG, RL, RP as per guide 2) Added Local Route Group to Device Pools Generally speaking, when a lab question states that a call is to be routed out of the Local gateway that is a clue that they want you to use the Standard Local Route Group. In this case, you would create a route group (rg-hq) for the HQ router (10.10.200.3). This is the RG that is placed in the HQ Device Pool. You then create a Route List (rl-local) and select Standard Local Route Group. The 911 route pattern will use the rl-local in the Gateway/Route List selection box. Again, double check that you have all of the above correct. Another item to check is the CSS set on the HQ phone, but that's probably not the issue here, as you already stated that you have the 911 patern in the none partition. When you say that there is no debug, what debug commands are you refering to? Debug isdn q931 ? This will show call atempts out of the HQ router toward the PSTN. Also do a debug voip dialpeer as this will verify if the call is making its way into the router and what dial-peers it's attempting to use. Make sure you have an inbound dial-peer configured and are not relying on dial-peer 0. On Thu, Oct 27, 2011 at 11:33 AM, Voiper datapack...@gmail.com wrote: Greetings to all: I seek help from those who have tread the path. Workbook Volume 1, lab 5.2 and have am already stuck :( Followed the PG and the walk through video with little success. Question 5.2 - All calls from UCM phones to Emergency Services must be routed out of the Local gateway. - Emergency Services can be dialed by entering 9-1-1 - The ANI should be in full E168 format - the +, country code and the national digits should be sent to the PSTN - You should configure the phones such that the telephone number in the top right of the screen of the phone displays the full E164 number 1) Created the RG, RL, RP as per guide 2) Added Local Route Group to Device Pools 3) voice translation-rule 911 rule 1 /^1/ /+1/ ! voice translation-profile ANI-OUT translate calling 911 4) dial-peer voice 911 pots translation-profile outgoing ANI-OUT destination-pattern 911 port 0/0/0:23
Re: [OSL | CCIE_Voice] RSVP
The CUCM is most likely trying to use g711 since the Device Pool of the MTP's are incorrect. Make sure: - HQ phone is in DP-HQ - HQ MTP is in DP-HQ - BR1 phone is in DP-BR1 - BR1 MTP is in DP-BR1 Codec passthru is not required if you have set codec g729r8 (assuming g729 is being used over the WAN). So that should be ok. On Oct 28, 2011, at 4:05 AM, Robert Schuknecht wrote: Hi, i would say double check your region settings. It seems that cucm wants to use G711 codec. Inside your dspfarm, you are missing the codec passthrough command. Is this by accident? As far as I know the RSVP locations, inside Callmanager, should have unlimited BW assigned. But, first of all I would double check the Region Settings inside CUCM. Do not rely on the Service Parameter Settings for Inter/Intra Region Codec, I saw some issues there in real live (cucm 7/8). /Robert Von: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftrag von DeShon Crayton Gesendet: Freitag, 28. Oktober 2011 03:30 An: ccie_voice@onlinestudylist.com Betreff: Re: [OSL | CCIE_Voice] RSVP Hello, I can working on a lab that requires RSVP. My setup is as follows: IOS 12.4.15(T14) UCM 7.0.1 ***HQ*** interface Virtual-Template200 bandwidth 768 ip address 192.168.1.1 255.255.255.252 ppp multilink ppp multilink interleave ppp multilink fragment delay 10 service-policy output br1 ip rsvp bandwidth 64 interface Virtual-Template200 bandwidth 768 ip address 192.168.1.1 255.255.255.252 ppp multilink ppp multilink interleave ppp multilink fragment delay 10 service-policy output br1 ip rsvp bandwidth 64 ***BR1*** interface Virtual-Template200 bandwidth 768 ip address 192.168.1.2 255.255.255.252 ppp multilink ppp multilink interleave ppp multilink fragment delay 10 service-policy output AutoQoS-Policy-UnTrust ip rsvp bandwidth 64 dspfarm profile 1 mtp codec g729r8 rsvp maximum sessions software 2 associate application SCCP I have the generic Hub_None location and a location for BR1. The bandwidth allowed for BR1 is 48K(per UCM). This will accommodate (2) g729 calls. I have a region for HQ and a region for BR1. The inter-region codec is g729. I am trying to figure of why the RSVP reservation request is being sent out for 96K? Oct 28 01:05:30.493: RSVP 172.16.12.1_18400-172.16.22.1_18508[0.0.0.0]: start requesting 96 kbps FF reservation for 172.16.12.1(18400) UDP- 172.16.22.1(18508) on Virtual-Access3 neighbor 192.168.1.1 I understand that per the UCM SRND, g711 calls initially request 96K. The initial request for a g729 call should be 40K. Can anyone shed any light on my issue? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CUE-2-CCM Can't dial VM 3600
Are CTI Rps and CTI ports registered? When you sh run inside the cue cli do you see the CTI ports under ccn subsystem jtapi? Also check the ccn trigger jtapi phonenumber Do you hear annunciatior or fast busy? More info pls -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 On Oct 28, 2011, at 5:29 PM, edgar feliz ejfeli...@yahoo.com wrote: I have set up CUE to CCM configured CTI ports, vm profile etc. double checked with all the IE guide and can't get BR2 phone to dial into VM? Any Ideas? Users imported int0 CUE OK, CCM can see the 10.10.202.2 CUE when I look at CTI RPs. EJF ___ 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] AAR
Location can always be assigned at the Device Pool level for both the phones and gateways. It is not necessary to set the Location at the device/gw level. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Oct 24, 2011, at 4:43 PM, Santiago Figueroa wrote: Hello Experts I have a question, when using AAR group the Location assignment for each one IP phones and gateways or only is necessary to add in device pool? Thanks, La información incluida en este mensaje y sus anexos es CONFIDENCIAL y para USO EXCLUSIVO de sus destinatarios. No está permitida su divulgación y/o reproducción sin autorización. Si ha recibido este mensaje y no le incumbe, le rogamos nos los comunique y proceda a su borrado. Gracias. Information included in this e-mail and attached files is CONFIDENTIAL and only for the EXCLUSIVE USE of the receivers. Circulation and/or copy without permission is not allowed. If you have received this e-mail and you are not the intended recipient, please let us know and erase the message and attached files. Thank you. ___ 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] Vol2 Lab6: MVA by MGCP hairpin outbound calls don't work
Its easier to make the DID number you call for MVA and the MVA DN a DIFFERENT number. Also you have a codec problem. Keep the DID# 5010. Change the MVA DN 5011. This is under the Media Resource menu in the ccmadmin page. Change the dial-peer to look like this: dial-peer voice 5 voip service cmm destination-pattern 5011 equal to MVA DN session target ipv4:10.10.210.10 incoming called-number 5010 equal to DID dtmf-relay h245-alphanumeric voice-class codec 1 you need to support 729 and 711 since you are making a call over the WAN no vad Make sure that you have the h323-g voip bind src in the interface you are using when you add the H323 gw. Reset the H323 gw too. Keep the MVA DN in the None partition. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Oct 24, 2011, at 10:18 PM, Anthony Alba wrote: Hello, When I try to use MVA with MGCP hairpin I cannot make calls. When I try to make a call I get the IVR menu again Dial 3945010 get IVR menu Enter PIN # Enter 1 1002# ( to make a call) ...instead of being connected I get back to the IVR menu. I seem to be trapped in some sort of loop. Any ideas? (When I change HQ-RTR to a H.323 gateway everything works including making calls. I think this means that my RDP CSS is looking good.) I have configured Mobile Voice Access as per the Solution Guide in Vol2 Lab6. HQ-RTR is running H.323 solely to provide VXML support. CUCM is configured to hairpin the call to HQ-RTR. application service cmm http://10.10.210.10:8080/ccmivr/pages/IVRMainpage.vxml dial-peer voice 5 voip service cmm destination-pattern 5010 session target ipv4:10.10.210.10 incoming called-number 5010 dtmf-relay h245-alphanumeric codec g711ulaw no vad ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] cue-cme mwi unsolicited question
The mwi-relay is required when you have phones registered to multiple CMEs with a mailbox in CUE. For your second question- if you have mwi sip in ephone-dn then you don't need unsolicited. But unsolicited would not break mwi if you did use it. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Oct 24, 2011, at 12:51 PM, zamuel del Toro wrote: O.k I read the blog and make me clear, I only have one more question?, when is used the mwi-relay on telephony-services and on voice register global mwi reg e64? because on unsolicited just need sip-ua mwi-server unsolicited and on subcribe notifiy sip-ua mwi-server ephone-dn mwi sip I notice to that when we have cucm-cue and phones are gone to srst the auto provitioning ephone-dn include mw sip even when cue is on unsolicited. that mean that cue must be subscribe notifiy and not unsolicited?. thanks everyone. Date: Sun, 23 Oct 2011 21:03:30 -0500 From: ash.ayy...@gmail.com To: ccielab...@gmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] cue-cme mwi unsolicited question Glad its all sorted out now , Thanks Ash On Sun, Oct 23, 2011 at 8:57 PM, ccielabrat ccielab...@gmail.com wrote: Hey Ashraf, You got me thinking the right way. I had a mismatch between my sip interface and the gateway configured on CUE. Thanks! On Sat, Oct 22, 2011 at 4:43 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote: did you binded the SIP to the correct interface from the CME config Voice service Voip ? Any chance to reload the Funky CUE ? Ash On Sat, Oct 22, 2011 at 12:12 PM, ccielabrat ccielab...@gmail.com wrote: I can't get CUE MWI working either. This is my cue config for SIP ccn subsystem sip gateway address 10.1.131.1 mwi envelope-info mwi sip unsolicited end subsystem I've tried all kinds of config on the CME router without success. When running debug ccsip messages on the CME router , I don't see anything if I issue mwi refresh all on CUE, even though I can dial into CUE and check to hear a voicemail on dn 4001 On Fri, Oct 21, 2011 at 6:47 PM, Brian btmulg...@gmail.com wrote: hi - this is an excellent summary of mwi for cue that is worth a read http://blog.ipexpert.com/2010/07/19/sip-mwi-mechansims-on-cue-notify/ Sent from my iPad On 21 Oct 2011, at 21:23, Ashraf Ayyash ash.ayy...@gmail.com wrote: Hello Zamuel , the mwi relay command is only needed in case of the subscribe notify MWI and its not needed in case of using Unsolicited because it does send the event to the phone using NOTIFY message no matter it subscribed to the MWI server Or not . Ash On Fri, Oct 21, 2011 at 11:21 AM, zamuel del Toro sdelto...@hotmail.com wrote: Hi Vic, how is it going?, about mwi unsolicited. sip-ua mwi.. unsolicited telephony-ser mwi relay ephone-dn nothing works mwi if subscribe notify sip-ua mwi... telephony-ser nothing ephone-dn mwi sip both works fine what if make mistake if on unsolicited include on ephone-dn mwi sip, that work too.is wrong do this? thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ 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
Re: [OSL | CCIE_Voice] problems with cucm and cu integration
Try calling from a local phone first (HQ/SA) to make things simpler. When you are routed to UC do you hear the correct subscriber greeting or are you hearing opening greeting? With the information you have provided it's difficult to progress this- but if you are leaving a message and it is not in UC then I would just call another phone and check that there is one-way audio problems when routing to UC. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Oct 13, 2011, at 4:34 AM, Zurdo Spike wrote: hi expert im having problems with the integration between CUCM and UC everything works fine but when i tried to leave a mess in the UC, the UC doesnt save the mess and then when i access to my voicemail i try to hear the mess but this doesnt appear. Voicemail ports are working. MWI is working. AXL is fine. Any ideas. ___ 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] SIP phones supp services on RAS GK trunk: revisited (WB2, Lab2)
I think we found this to be the firmware in the phone. Please confirm by performing a suppl service locally (remove GK from the equation). On Thu, Oct 13, 2011 at 12:45 AM, Anthony Alba ascanio.al...@gmail.comwrote: Hi folks, I read earlier on the list that some of you have got supplementary services with SIP phone working over the H.323 trunk. I am using 12.4(24)T5 with WB2 Lab2 task: CUCM to CME over a GK H.225 trunk. As per the earlier thread http://www.onlinestudylist.com/archives/ccie_voice/2011-February/072694.html I am just not getting the CME SIP phone to work with Hold, Transfer. The CME SCCP phone works fine. Phone calls are getting through, and I see the BR2 transcoder being invoked (and that's another story - why doesn't it just use G.729 to the SIP phone, like the SCCP phone). The CME SIP phone is able to invoke a call transfer to the CME SCCP phone, but that's about the only thing that is working. Any ideas? dial-peer voice 101 voip destination-pattern [15]... session target ras tech-prefix 1# dtmf-relay h245-alphanumeric no vad ! dial-peer voice 100 voip incoming called-number 3...$ dtmf-relay h245-alphanumeric no vad ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip sip bind control source-interface FastEthernet0/0.400 bind media source-interface FastEthernet0/0.400 registrar server telephony-service sdspfarm units 4 sdspfarm transcode sessions 8 sdspfarm tag 1 BR2-Xcode max-ephones 4 max-dn 20 ip source-address 10.10.110.3 port 2000 url authentication http://10.10.110.3/CCMCIP/authenticate.asp time-zone 43 time-format 24 date-format dd-mm-yy voicemail 3600 max-conferences 8 gain -6 transfer-system full-consult transfer-pattern .T after-hours block pattern 1 900 7-24 create cnf-files version-stamp 7960 Oct 12 2011 13:14:26 Cheers Anthony ___ 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 -- Vik Malhi – CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. ___ 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] Phoneview and CME
Little confusing. The application PHONEVIEW (as opposed to the CUC phoneview feature) is what we are talking about. It should work. Two things- check IOS is 12.4(22) and check the auth URL is downloaded to the phone (check device settings from phone). On Thu, Oct 13, 2011 at 9:18 AM, Mohammed Al baqari baqari.voic...@gmail.com wrote: As far as I know, Phone View isn't supported in CME integration with CUC. Only CUCM integration is supported. On Tue, Oct 11, 2011 at 6:32 PM, Emanuel Damasceno aedamasc...@gmail.comwrote: Phoneview doesn't seem to work with CME. Anyone having issues? I followed the tutorial on proctorlabs page, but I always get an error communicating with the server. I can ping the CME source-address ip from my machine just fine. Those were the configs I used... ip http server no ip http secure-server ! ! ixi transport http response size 64 no shutdown request outstanding 1 ! ixi application cme no shutdown ! ! ! telephony-service xml user admin password c1sc0123 15 url authe http://10.10.202.1/CCMCIP/authenticate.asp pview cisco *Antonio Emanuel Damasceno* CCNA, CCNA Voice, CCNP Voice, CCIE Voice (written) CompTIA Network+ ___ 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 -- Vik Malhi – CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. ___ 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] CCIE Voice Mobile Labs
I would have thought they would specifically mention that they have switched to remote phone management if this was the case On Wed, Oct 12, 2011 at 5:19 AM, Ken Wyan kew...@gmail.com wrote: Hi, Cisco has started CCIE Voice also mobile labs. https://learningnetwork.cisco.com/groups/ccie-voice-study-group That means ip phones are going to be virtual? (in other locations also) Ken ___ 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 -- Vik Malhi – CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. ___ 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] Mobile Labs Voice
Whether phones are virtualized or not should not in theory make a difference to the candidate. I would expect them to announce this officially so my guess is that physical phones will be present in the mobile testing centers. The other question is will they switch to electronic copies of labs from hardcopies? We have these two questions out to a few people who should be able to confirm either way (either on here or on Learning @ Cisco's site) shortly. Either way - it's no reason to panic and I would see no reason to doubt that Cisco have not thought about the issues that come with mobile testing centers. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Oct 12, 2011, at 4:45 PM, Emanuel Damasceno wrote: Hey guys, I saw the post of one of our brothers here about the Mobile Labs. I checked out that information and I saw that Mobile Labs will be in Buenos Aires in March of 2012. This will be a lot cheaper for me, since I am departing from Brazil. However, something bothers me. I saw Vik's post about the phones not being at the lab (physically present). At least it was what I understood. I looked everywhere to see if I could find that information, but I couldn't find it. Did I understand Vik correctly? I am thinking here that if that's true, I'd rather pay a little more and take the exam with the devil we already know. If the phones are not physically present, we are susceptible for more problems. Can anybody confirm? Thanks :) Antonio Emanuel Damasceno CCNA, CCNA Voice, CCNP Voice, CCIE Voice (written) CompTIA Network+ ___ 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] Problem with PhoneView
Regarding the config- looks ok to me. I have had problems with the phones running older firmware. I can control my 7965 registered to CME with the following firmware (which is got by registering to UCM 7.0) Version : SCCP45.8-4-1S Try this and let me know if you need further assistance. -- Vik Malhi CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ From: Abel ... midga...@gmail.com Date: Fri, 15 Jul 2011 18:11:02 -0400 To: OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Problem with PhoneView Anyone can help me with this? Thanks On Thu, Jul 14, 2011 at 8:41 PM, Abel ... midga...@gmail.com wrote: Hello everyone, I'm have this strange issue with PhoneView, for some reason registered phones can be remotely access from the App, a screenshot of the app is attached on the mail. Also this is using CME configuration, here is the config: Thanks for the help everyone BR2-RTR#sh running-config Building configuration... Current configuration : 3642 bytes ! ! Last configuration change at 02:33:50 UTC Fri Jul 15 2011 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname BR2-RTR ! boot-start-marker boot system flash c2800nm-adventerprisek9_ivs-mz.124-22.T5.bin boot-end-marker ! card type e1 0 0 logging message-counter syslog ! no aaa new-model clock timezone UTC 1 clock summer-time UTC recurring 1 Sun Apr 1:00 last Sun Oct 1:00 network-clock-participate wic 0 ! dot11 syslog ip source-route ! ! ip cef ip dhcp excluded-address 10.10.202.1 10.10.202.49 ip dhcp excluded-address 10.10.202.70 10.10.202.254 ip dhcp pool PHONES network 10.10.202.0 255.255.255.0 default-router 10.10.202.1 option 150 ip 10.10.110.3 ! ! no ipv6 cef ! multilink bundle-name authenticated ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! voice-card 0 ! ! ! ! ! archive log config hidekeys ! ! ! ! ! controller E1 0/0/0 ! ! ! ! ! interface Loopback0 ip address 10.10.110.3 255.255.255.255 ip ospf network point-to-point ! interface FastEthernet0/0 no ip address duplex auto speed auto ! interface FastEthernet0/0.200 encapsulation dot1Q 200 native ip address 10.10.102.1 255.255.255.0 ! interface FastEthernet0/0.400 encapsulation dot1Q 400 ip address 10.10.202.1 255.255.255.0 ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/1/0 no ip address encapsulation frame-relay IETF no fair-queue frame-relay lmi-type ansi ! interface Serial0/1/0.1 point-to-point ip address 10.10.112.2 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 102 ! interface Service-Engine1/0 no ip address shutdown ! router ospf 1 router-id 10.10.202.1 log-adjacency-changes network 10.10.0.0 0.0.255.255 area 0 ! ip forward-protocol nd ip http server no ip http secure-server ! ! ! ! ixi transport http response size 4 no shutdown request outstanding 1 ! ixi application cme no shutdown ! ! ! ! ! ! ! control-plane ! ! ! ccm-manager fax protocol cisco ! mgcp fax t38 ecm ! ! ! ! ! ! gatekeeper shutdown ! ! telephony-service xml user admin password cisco 15 max-ephones 4 max-dn 4 ip source-address 10.10.110.3 port 2000 auto assign 1 to 4 url authentication http://10.10.110.3/CCMCIP/authenticate.asp admin cisco network-locale ES network-locale 1 ES network-locale 2 ES network-locale 3 ES network-locale 4 ES max-conferences 8 gain -6 transfer-system full-consult server-security-mode non-secure create cnf-files version-stamp 7960 Jul 15 2011 02:21:16 ! ! ephone-dn 1 dual-line number 3001 ! ! ephone-dn 2 dual-line number 3002 ! ! ephone-dn 3 dual-line number 3003 ! ! ephone-dn 4 dual-line number 3004 ! ! ephone 1 no phone-ui speeddial-fastdial no multicast-moh device-security-mode none mac-address 8CB6.4FF7.EB14 keepalive 30 auxiliary 0 codec g729r8 pre-ietf type 7965 button 1:1 ! ! ! ephone 2 no phone-ui speeddial-fastdial no multicast-moh device-security-mode none mac-address 40F4.ECEE.6924 keepalive 30 auxiliary 0 codec g729r8 pre-ietf type 7965
Re: [OSL | CCIE_Voice] CUBE and codec
It all depends on whether Early Offer is being used or not. And whether UCM is originating the call. Can you explain what is either side of the CUBE? If UCM is the point of origination and Early Offer has been selected then an MTP is going to be used and the codec being used by the MTP is most likely the problem. -- Vik Malhi CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ From: Ali El Moussaoui mousawi@gmail.com Date: Wed, 13 Jul 2011 22:07:05 +0300 To: OSL Group ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CUBE and codec Hello, How can i enforce the codec choice on in and out legs with a sip-sip scenario call? when looking into debug output i see the codec slection is different that the codec class. This matter is giving me hard time and my calls are all failing. Any suggestions would be highly appreciated. Note: I am thinking of changing the IOS what do u recomment for a SIP-SIP cube? Ali ___ 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] srst calling name
You have to use auto-provision all or dn and then change the name in the learned ephone dn's. Then take the site out of SRST and save. -- Vik Malhi CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ From: mgscip gpsvoiceexpe...@yahoo.com Reply-To: mgscip gpsvoiceexpe...@yahoo.com Date: Wed, 13 Jul 2011 11:35:08 -0700 (PDT) To: OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] srst calling name HI Experts , Please advice on this. From: donny f f.faraday...@gmail.com To: ccie_voice@onlinestudylist.com Sent: Monday, July 11, 2011 11:27 AM Subject: [OSL | CCIE_Voice] srst calling name hi all, Assuming we been told to send calling name SiteC ph 2 in SRST mode call to PSTN. How do we achieve it it, if we ask to only use srst autoprovision all and srst autoprovision dn cause evertyme after u change Name to SiteC ph 2 from +6132 When you switch to UCM mode and back to SRST, it will overide tks for advice ___ 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] srst calling name
No- the config is saved and the existing ephones and dns will be reused when the phones fall back into srst. -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ On Jul 14, 2011, at 8:46, donny f f.faraday...@gmail.com wrote: Vik and others, After change name ,out of SRST,and save. DO I need to put srst auto-provision-none under telephony-service, so next time when they fallback to SRST (it will not override) ? On Thu, Jul 14, 2011 at 2:36 AM, Vik Malhi vma...@ipexpert.com wrote: You have to use auto-provision all or dn and then change the name in the learned ephone dn's. Then take the site out of SRST and save. -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ From: mgscip gpsvoiceexpe...@yahoo.com Reply-To: mgscip gpsvoiceexpe...@yahoo.com Date: Wed, 13 Jul 2011 11:35:08 -0700 (PDT) To: OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] srst calling name HI Experts , Please advice on this. From: donny f f.faraday...@gmail.com To: ccie_voice@onlinestudylist.com Sent: Monday, July 11, 2011 11:27 AM Subject: [OSL | CCIE_Voice] srst calling name hi all, Assuming we been told to send calling name SiteC ph 2 in SRST mode call to PSTN. How do we achieve it it, if we ask to only use srst autoprovision all and srst autoprovision dn cause evertyme after u change Name to SiteC ph 2 from +6132 When you switch to UCM mode and back to SRST, it will overide tks for advice ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
