[OSL | CCIE_Voice] quto qos voip trust in WAN
hi all, my understanding is when we used quto qos voip trust in HQ router , because we trust classification from SWITCH that connect to HQ router. And in SB router, we can't use quto qos voip trust but auto qos voip only because SB site do not have SWITCH. is this understanding correct ? tks 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
Re: [OSL | CCIE_Voice] B-ACD
Thanks somphol i wll try that and let you know On Sun, Jun 23, 2013 at 11:56 AM, Somphol Boonjing somp...@gmail.comwrote: Remove all of the param under the service did the trick, Say you have this in your running config application service app-b-acd param number-of-hunt-grps 2 param aa-hunt1 param aa-hunt2 1222 param queue-len 15 param queue-manager-debugs 1 ! Then, application service app-b-acd no param number-of-hunt-grps 2 no param aa-hunt1 no param aa-hunt2 1222 no param queue-len 15 no param queue-manager-debugs 1 Once there is no param set for the service, it will be removed from the running-config. --- Detail trace below: --- Branch2#show run | begin application application service app-b-acd param queue-len 15 param aa-hunt1 param queue-manager-debugs 1 param aa-hunt2 1222 param number-of-hunt-grps 2 ! ! Branch2(config)#application Branch2(config-app)# service app-b-acd Branch2(config-app-param)#no param queue-len 15 Warning: parameter queue-len has not been registered under app-b-acd namespace Branch2(config-app-param)#no param aa-hunt1 Warning: parameter aa-hunt1 has not been registered under app-b-acd namespace Branch2(config-app-param)# Branch2(config-app-param)#do show run | begin application application service app-b-acd param queue-manager-debugs 1 param aa-hunt2 1222 param number-of-hunt-grps 2 ! ! Branch2(config-app-param)#no param queue-manager-debugs 1 Warning: parameter queue-manager-debugs has not been registered under app-b-acd namespace Branch2(config-app-param)#no param aa-hunt2 1222 Warning: parameter aa-hunt2 has not been registered under app-b-acd namespace Branch2(config-app-param)#no param number-of-hunt-grps 2 Warning: parameter number-of-hunt-grps has not been registered under app-b-acd namespace Branch2(config-app-param)#do show run | begin application associate application SCCP ! dspfarm profile 5 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 On Sun, Jun 23, 2013 at 12:20 AM, Bill Lake whl...@gmail.com wrote: Try doing all command not just these Sent from my iPhone On Jun 22, 2013, at 6:51 AM, CISCO CCIE VOICE ccievoic...@gmail.com wrote: Thanks Bill for your reply, I have done no service app-b-acd and no service app-b-acd-aa but showing all those commands in Running configuration thanks On Sat, Jun 22, 2013 at 1:48 PM, Bill Lake whl...@gmail.com wrote: If it is showing up in the running configuration, then you most likely see something like below, the best way to remove this is to no the commands Or to have done a Archive or copy of the config before you apply it. then restore that config as the startup and reboot. application * service app-b-acd * param number-of-hunt-grps 2 param aa-hunt2 param aa-hunt3 1222 param queue-len 15 param queue-manager-debugs 1 ! * service app-b-acd-aa * paramspace english index 1 paramspace english language en paramspace english location flash: param service-name app-b-acd param handoff-string app-b-acd-aa param aa-pilot 8005550123 param welcome-prompt _bacd_welcome.au param number-of-hunt-grps 2 param dial-by-extension-option 1 param second-greeting-time 60 param call-retry-timer 15 param max-time-call-retry 700 param max-time-vm-retry 2 param voice-mail 5003 ! dial-peer voice 222 voip service app-b-acd-aa destination-pattern 8005550123 session target ipv4:192.168.1.1 incoming called-number 8005550123 dtmf-relay h245-alphanumeric codec g711ulaw no vad On Sat, Jun 22, 2013 at 2:25 AM, Somphol Boonjing somp...@gmail.comwrote: That one is the embedded one so you actually can not remove it. However, you can simply ignore it and use one that is external script. So, if you have the external BACD script, you can use it instead of the embedded one. Branch2#show flash | inc bacd 107 30421bacd/app-b-acd-3.0.0.2.tcl 108 55599bacd/app-b-acd-aa-3.0.0.2.tcl application service *funnyqueue flash:/bacd/*app-b-acd-3.0.0.2.tcl -- you can you whatever name you like, in this case funnyqueue -- point the script to the script with correct path . (detail remove for brevity)... ! service* funnyaa flash:/bacd/app-b-acd-aa-3.0.0.2.tcl * -- you can you whatever name you like, in this case funnyaa -- point the script to the script with correct path . (detail remove for brevity). param service-name *funnyqueue* -- refer to your queue application name param handoff-string *funnyaa* . (detail remove for brevity). ! dial-peer voice 222 voip service *funnyaa* -- refer to your AA application name. . (detail remove for brevity)... !
Re: [OSL | CCIE_Voice] CCIE Collaberation
Anyone attending cisco live can update rest of us ? On Thu, Jun 6, 2013 at 5:32 PM, Ken Wyan kew...@gmail.com wrote: Finally Cisco understood the mistake corrected it before further complications begin. One guy passed RS v.2 lab later passed RS v.3 lab finally passed RS v.4 lab. He is a true expert in RS but not counted as a dual / triple CCIE for partner status. Now cisco had to offer dual / triple / quadruple CCIE status for single track CCIE s as well. Before things get complicated , luckily Cisco corrected the mistake. On Thu, Jun 6, 2013 at 5:57 PM, khaled Saholy khaled_sah...@hotmail.comwrote: This is really awesome news. Now we can go on with more enthusiasm. Thank you all. Khaled -- Date: Thu, 6 Jun 2013 12:35:01 +0300 Subject: Re: [OSL | CCIE_Voice] CCIE Collaberation From: aryan231...@gmail.com To: khaled_sah...@hotmail.com yes! cisco approved to ccie voice to become collaboration with written exam... On Wed, Jun 5, 2013 at 9:25 PM, khaled Saholy khaled_sah...@hotmail.comwrote: Hi friends, Any news on our subject!! Regards. khaled -- From: hugo.barr...@nexusis.com To: whl...@gmail.com Date: Tue, 4 Jun 2013 15:12:43 + CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE Collaberation Well spoken Bill 100% agree. Regards, Hugo On Jun 3, 2013, at 3:58 PM, Bill Lake whl...@gmail.com wrote: Right you can continue to pass written exams and prove you are current with technology. But what will you do when you apply for a job that requires or desires CCIE Collaboration? CCIE voice will join the CCIE's of the pass that were retired, you will still be a CCIE but you will be considered less important because your CCIE is on outdated technology, like CCIE ISP Dial or CCIE SNA/IP integration do you see anyone asking for them now? So I would prefer to have a grandfather or upgrade path instead of having to take the CCIE Collaboration exam For those that say why not just get dual CCIE with CCIE Collaboration as it should be easy, name the people who say their CCIE was easy and would gladly pay 1500 more to take it again just to renew their current CCIE. No if I want to get a dual CCIE, I want to earn it in another track, not another voice or communications centric track Bill On Mon, Jun 3, 2013 at 1:09 PM, Leslie Meade leslie.me...@lvs1.com leslie.me...@lvs1.com wrote: Wonder what this means…. ** ** http://ciscocert.force.com/english/articles/Article/CCIE-Collaboration-Certification-CCIE-Voice-can-be-recertified-EN?retURL=/english/apex/instantanswers?productCategory%3DCCIE_collaborationpopup=false http://ciscocert.force.com/english/articles/Article/CCIE-Collaboration-Certification-CCIE-Voice-can-be-recertified-EN?retURL=%2Fenglish%2Fapex%2Finstantanswers%3FproductCategory%3DCCIE_collaborationpopup=falsehttp://ciscocert.force.com/english/articles/Article/CCIE-Collaboration-Certification-CCIE-Voice-can-be-recertified-EN?retURL=/english/apex/instantanswers?productCategory%3DCCIE_collaborationpopup=false ** ** ** ** The CCIE Collaboration certification does not directly affect current CCIE Voice certification holders. Current CCIE Voice holders will be able to recertify by passing any CCIE exam including the new CCIE *Collaboration written or lab exams*. The CCIE Collaboration certification provides new career opportunities for CCIE Voice certification holders ** ** ** ** *Leslie Meade* image001.jpg Bach Information Technology CCNA CCVP CCIE Voice 38727 Network Consultant .. * Mobile:778.228.4339* | *Main:* *604.676.5239* *Email:* leslie.me...@lvs1.com image002.jpg http://www.linkedin.com/company/17908 image003.jpghttp://twitter.com/LongViewSystems image004.jpg http://www.facebook.com/longviewsystems image005.jpghttp://gplus.to/longviewsystems image006.jpg http://www.youtube.com/longviewsystemsimage007.gif www.longviewsystems.com This message and any attached documents are only for the use of the intended recipient(s), are confidential and may contain privileged information. Any unauthorized review, use, retransmission, or other disclosure is strictly prohibited. If you have received this message in error, notify the sender immediately, and delete the original message.*** * ** ** ** ** ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.comwww.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out http://www.PlatinumPlacement.com www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.comwww.ipexpert.com Are you a CCNP or CCIE and looking for a job?
Re: [OSL | CCIE_Voice] Codec and CAC section
This may have nothing to do with the implementation checklist, but may be useful as part of a verification steps. Assuming SiteA SiteC are configured with a typical scenario with G711 inter-region and G729 intra-region, and RSVP is required. 1. Incoming or outgoing PSTN call to or from say SC Phone 1. Verify the codec of the active call using ?? that it is G711.Put the call on Hold. Assuming the MoH is in SiteA. Is it successful? Does music going through to the PSTN phone successfully? 2. Incoming or outgoing PSTN call to or from say SC Phone 1. Verify the codec of the active call using ?? that it is G711. Transfer the call to SA Phone 1. Is it successful? If yes, then from SA Phone 1, put the call on Hold.Is it successful? 3. Make a call from SC Phone 1 SA Phone 1, verify on both phone that the codec for the active call is G729.Verify on the gateway using the following commands to spot anything obviously problem, such as 80K is reserved instead of 24K, or G729abr8 or G729ar8 is used instead of G729, etc. # show ip rsvp interface # show ip rsvp installed # show sccp connections # show sccp connections rsvp # show sccp connections detail 4. (optional) Set CFWDALL on SC Ph1 to SA Ph1 DN. Make PSTN call to SC Ph1 DN, the call should ring SA Ph1.Answer the call, verify the active codec, etc. This is just my idea of a partial verification steps that may help isolate any problem in your configuration. Another thing is that I find useful for my study. I often create the following MTP for each site, put them in different MRG so that I can add/remove/re-order them in MRGL to see how CUCM select different resources. In case you find it useful too. - MTP: G711 only + RSVP - MTP: G729 only + RSVP - MTP: Pass-through only + RSVP - XCODER: XCODER + RSVP Hope this doesn't deviate too much from your question. Regards, --Somphol On Mon, Jun 24, 2013 at 4:26 AM, Karen Johnson karen.johnson...@yahoo.cawrote: hi folks, can anyone share experience on what to check on this section , I got 0 for few attempt. Here is what I did : UCM = - service parameter : no G722 and ILBC - Enterprise parameter G711 intra, G729 inter - Region : HQ SB SC, HQ-HQ : G711 , SB-SB G711, SC-SC : g711 (rest relation : G729) and assign tp DP - Location : HQ and SC : mandatory , assign to DP - MRGL HQ -- MRG-- MTP from HQ same for SC , assign to DP router HQ and SC = - dspfarm profile 3 mtp codec pass-through codec g729r8 rsvp maximum sessions software 4 (as they asked 4 session of g729) associate application SCCP - interface Serial0/0/0.1 point-to-point frame-relay interface-dlci 102 ip rsvp bandwidth 112 verification = - call hq to hq, sb sb : g711, inter site phone and GW : g729 - sh ip rsvp reservation : 40 k (ring) , and 24 k (connect) question: - did i miss something critical that cause the mark to be 0 ? ___ 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] Codec and CAC section
hi Somphol, thanks for your advice here. could you pls help me to undertsand the concern when we check this ? - Assuming the MoH is in SiteA. Is it successful? Does music going through to the PSTN phone successfully? ( do Cisco always expect G711 when they did not say in exam ? ) - Transfer the call to SA Phone 1. Is it successful? If yes, then from SA Phone 1, put the call on Hold. Is it successful? (do you mean to chekc if MOH g729 here?) - (optional) Set CFWDALL on SC Ph1 to SA Ph1 DN. Make PSTN call to SC Ph1 DN, the call should ring SA Ph1. Answer the call, verify the active codec, etc. (this should G729,right ?) - From: Somphol Boonjing somp...@gmail.com To: Karen Johnson karen.johnson...@yahoo.ca Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Monday, June 24, 2013 5:42:26 AM Subject: Re: [OSL | CCIE_Voice] Codec and CAC section This may have nothing to do with the implementation checklist, but may be useful as part of a verification steps. Assuming SiteA SiteC are configured with a typical scenario with G711 inter-region and G729 intra-region, and RSVP is required. 1. Incoming or outgoing PSTN call to or from say SC Phone 1. Verify the codec of the active call using ?? that it is G711. Put the call on Hold. Assuming the MoH is in SiteA. Is it successful? Does music going through to the PSTN phone successfully? 2. Incoming or outgoing PSTN call to or from say SC Phone 1. Verify the codec of the active call using ?? that it is G711. Transfer the call to SA Phone 1. Is it successful? If yes, then from SA Phone 1, put the call on Hold. Is it successful? 3. Make a call from SC Phone 1 SA Phone 1, verify on both phone that the codec for the active call is G729. Verify on the gateway using the following commands to spot anything obviously problem, such as 80K is reserved instead of 24K, or G729abr8 or G729ar8 is used instead of G729, etc. # show ip rsvp interface # show ip rsvp installed # show sccp connections # show sccp connections rsvp # show sccp connections detail 4. (optional) Set CFWDALL on SC Ph1 to SA Ph1 DN. Make PSTN call to SC Ph1 DN, the call should ring SA Ph1. Answer the call, verify the active codec, etc. This is just my idea of a partial verification steps that may help isolate any problem in your configuration. Another thing is that I find useful for my study. I often create the following MTP for each site, put them in different MRG so that I can add/remove/re-order them in MRGL to see how CUCM select different resources. In case you find it useful too. - MTP: G711 only + RSVP - MTP: G729 only + RSVP - MTP: Pass-through only + RSVP - XCODER: XCODER + RSVP Hope this doesn't deviate too much from your question. Regards, --Somphol On Mon, Jun 24, 2013 at 4:26 AM, Karen Johnson karen.johnson...@yahoo.ca wrote: hi folks, can anyone share experience on what to check on this section , I got 0 for few attempt. Here is what I did : UCM = - service parameter : no G722 and ILBC - Enterprise parameter G711 intra, G729 inter - Region : HQ SB SC, HQ-HQ : G711 , SB-SB G711, SC-SC : g711 (rest relation : G729) and assign tp DP - Location : HQ and SC : mandatory , assign to DP - MRGL HQ -- MRG-- MTP from HQ same for SC , assign to DP router HQ and SC = - dspfarm profile 3 mtp codec pass-through codec g729r8 rsvp maximum sessions software 4 (as they asked 4 session of g729) associate application SCCP - interface Serial0/0/0.1 point-to-point frame-relay interface-dlci 102 ip rsvp bandwidth 112 verification = - call hq to hq, sb sb : g711, inter site phone and GW : g729 - sh ip rsvp reservation : 40 k (ring) , and 24 k (connect) question: - did i miss something critical that cause the mark to be 0 ? ___ 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/ - MTP: G711 only + RSVP - MTP: G729 only + RSVP - MTP: Pass-through only + RSVP - XCODER: XCODER + RSVP (may i know what is purpose) - if they ask codec G711, we should see 80k in sh rsvp reservation ? tks 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
Re: [OSL | CCIE_Voice] Codec and CAC section
Hi Karen, ( do Cisco always expect G711 when they did not say in exam ? ) My short answer is I don't know. I have seen a statement on record along that line of Don't assume what is not stated, however. The exam grading process seems to be a black box. There are guidelines such as Don't assume what is not stated and Every word is there for a reason.And being mortal myself, I haven't transformed to master the Zen of those words, hence I am struggling with a lot of contradiction. Sometimes, people seems to say Think about the implication and When there are a few ways to achieve certain outcome, some of them may not be desirable as it may break other requirement.It is confusing isn't it, when you must not assume but at the same time you need to think outside the box.How on earth there is such a thing!!! Assuming that all of those are true, we just need to find a way out the misery and hope that the contradiction is just that, a perception.All the good puzzles are hard when you try it, only when you master it that you see that it actually makes sense.And, we both are in that process of being transformed mentally -- to possess the mindset of a capable CCIE. I am on the journey myself, while I want to share what I think and hope it help, pick what is logical to you.Who know, I may have to take 20 attempts myself. And, when I actually pass, I may not be able to tell you what I actually see, as that will violate the NDA. My thought is that a simple requirement in the exam may be achieved by X different choices of configurations. Some of those configurations may be correct and some of those will (or may) cause other requirement to break -- outright or randomly -- when certain triggers happen such as when a PUB is down or when a WAN is down, etc. Within a rush of 8 hours, I think it is highly possible that we would fail to see that the choice we chose is indeed break other requirements. In some section, a checklist would help, in other section, however, we may need to dig a lot deeper as a simple change in requirement means different required configuration. - Transfer the call to SA Phone 1. Is it successful? If yes, then from SA Phone 1, put the call on Hold.Is it successful? (do you mean to chekc if MOH g729 here?) Check if the music is heard on PSTN phone. Check the audio codec of the active stream to see whether it meets the requirements. Most likely if the configuration is wrong, you won't be able to hear the music. Call Transfer Call Forward across Region are known to involve a complex set of underlying operations. That's why I think it is a good test case. - (optional) Set CFWDALL on SC Ph1 to SA Ph1 DN. Make PSTN call to SC Ph1 DN, the call should ring SA Ph1.Answer the call, verify the active codec, etc. (this should G729,right ?) Yes, in this scenario. - MTP: G711 only + RSVP - MTP: G729 only + RSVP - MTP: Pass-through only + RSVP - XCODER: XCODER + RSVP (may i know what is purpose) When I look at sample configurations, there are something that doesn't add up for me. The odd one was codec pass-through. (i.e. There is no clarification as per when this particular type of MTP is needed in its own right. Why does it need to exist in the sample config. What if I remove it. What if, etc) So, I use the above idea to experiment. And, when you do you lab, play around with it, you will discover a lot of weird outcomes initially, but once you understand it fully, it will make sense. I use RTMT to keep track of available media resources and active media resources. And, make a few calls here and there such as what I suggested above. There are two bugs you will need to be aware of concerning the version of CUCM IOS in lab blueprint. [1] MoH interaction with H323 gateway for PSTN calls (the workaround is to use MTP Required.) [2] CSCsl74701 ( http://ciscovoiceguru.com/382/cscsl74701-bug-details/) which is not directly about RSVP, but I have seen a case where 80K is allocated between my Pass-through+RSVP MTPs while G729 codec is active in a call between two phones in different regions. (The workaround is to set both inter-region intra-region codec to G729 and explicitly set the Region matrix codec for intra-region to G711.) The following example shows an IOS-based MTP with three capabilities -- able to support G729r8, able to talk RSVP, and able to support blindly tunnel the RTP data blindly (hence the term codec pass-through).If you mixed them up in one MTP, you will never know why certain MTP is chosen. So, I break them up and experiment and from RTMT I can tell which Media Resources are involved in certain scenario and try to make sense of it. dspfarm profile 2 mtp no codec g711u codec g729r8 codec pass-through rsvp maximum sessions software 100 associate application SCCP! So, I created a few MTPs for each site as shown above.Initially I put everyone of them in the same MRG and it is
