Re: [OSL | CCIE_Voice] [CORRECTED] CUE QOS on the Switch
thanks bill !! And thats why we call it state of the art, greatest, cutting edge Catalyst :) .. honor my sarcasm...i spent couple of hours playing with the accesslist, policymaps etc ... From: b...@ucguerrilla.com Date: Wed, 17 Apr 2013 19:08:11 -0400 To: kamina.j...@yahoo.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] [CORRECTED] CUE QOS on the Switch Yeah, on the catalyst switch you won't see the counters increment. That's normal. I would tweak the policy-map: policy-map CUEMAP class CUE-SIGNAL set ip dscp cs3 police 32 8000 exceed-action policed-dscp-transmit!You also need to modify your policed-dscp map: mls qos map policed-dscp 24 to 0 -Bill --William Bell, CCIE #38914blog: http://ucguerrilla.comtwitter: @ucguerrilla On Apr 17, 2013, at 5:59 PM, Jack Kamina wrote:on one of the practice lab the need is to police the signaling packets to and from CUE inbound into the HQ switch to 32 kbps and then remark the DSCP to 0. I built up the config below but dont see any packets matched on the show policy-map interface command. CUE IP is 10.1.6.253 . CUCM IP is 10.10.210.10 (pub) and 10.10.210.11 (sub) .is the access list built correctly? access-list 110 permit tcp host 10.1.6.253 any eq 2748 ! class-map match-all CUE-SIGNAL match access-group 110 ! policy-map CUEMAP class CUE-SIGNAL set dscp af31 bandwidth 20 ! interface Fa0/1/0 description HQ-ROUTER-INTERFACE service-policy input CUEMAP mls qosmls qos map cos 0 8 16 24 32 46 48 56mls qos map policed-dscp 24 26 to 8 ___ 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 functionality
And where do you plan to invoke the script and vxml function? Date: Wed, 27 Mar 2013 23:51:10 +0300 From: aboaz...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MVA functionality Hello Friends... I have the following setup, I am not sure if the will be suitable to enable the MVA feature ! I have CUCM cluster, but his CUCM cluster has no voice GW or DID .. but this CUCM cluster has Inter-cluster trunk to another CUCM cluster which has the DID numbers ? Can I configure the MVA for this setup.. Appreciate your input. ___ 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] clock summer-time command
of the list of the commands that dont make much sense, my fav is the clock summer-time .. lots of CCIE experts who passed have said they did the summer time command .. does HK have summer time and do we need to put this command on HK as well? ... in Bahrain we dont have summer, winter, monsoon times .. we follow 1 time :) ..time to take some geography lessons before embarking the CCIE adventures. 👲 my question: is this command absolutely necessary? PIXAR ___ 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 IN 1 Hr
grand welcome to the lab 7 carnival! many people get the lab7 on 2nd attempt , do the same things and score very high points and fail or pass (no visibility to points) .. so with this lab 7 there is no definitive answer on many things like CUPC, signaling, call routing. Date: Sun, 24 Mar 2013 01:33:10 +0530 From: peterrodyc...@gmail.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] LAB IN 1 Hr Got lab 7 in SJC very big lab i completed in 6 hrs time was having full hopes will clear the test Lost in few sections like VG i got only 53% and HA less marks and did not got marks in LAN qos. I did exactly the way cisco expected this is really bugging me about VG section as it was total 17 marks If anyone want to be study partner please email me so that we can crack the lab my friend attempted the same day he got lab 7 again twice its really really painful because we both got 53% in VG and i want to know why all people missing in this for no reasons. Waiting for all to reply, Email me so that we can talk more rest all sections i got 100% so if needed i can share the solutions. On Sat, Mar 23, 2013 at 3:57 AM, Josh Petro wrote: Good luck and God Bless! On Mar 22, 2013 10:39 AM, "Peter Rody" wrote: Hello friends, Ready for the attempt just starting from hotel now its long practice please pray for my attempt. Everyone comes out say i pass its feel so good but i think i might be first guy need your prayers starting from my hotel now. Not able to sleep whole night properly. Feel like on a top of the world now Thanks zillon to all of 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 ___ 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 to voicemail without Alternate Extension
the requirement is always for SiteB calling into SiteA voicemail by hitting Messages button. SiteA is always MGCP gateway. Calling Party Xform isnt any use on MGCP gateway for incoming calls. here is another way of doing it ... Voicemail Pilot for CUC is 2200 call-manager-fallbackvoicemail 2777 ---> siteB specific translation-pattern on CUCM to convert 2777 into 2200 and mask calling number . The CSS of the translation pattern should have access to 2200. there is no definitive answer as to which solution is graded positively. there is a reason why many leading CCIE instructors say this is not a test of best practices but a test of how like able is your solution to the script. .. :) From: marth...@cisco.com To: skeller...@gmail.com Date: Thu, 21 Mar 2013 03:59:48 + CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension What about a calling party transform mask on the incoming gateway? Sent from my iPhone On Mar 20, 2013, at 10:43 PM, "Steve Keller" wrote: Thanks Bill, I like this option pretty well as it seems to limit treatment of calls this way to CUC when site B is in SRST mode only. I will try to lab this up tomorrow morning. Question for you, will this only solve my issue of pressing the VM button to access my mailbox to retrieve a message. Meaning when PSTN calls in to site B phone and then gets forward(redirected) to voicemail, I use a dial-peer that provides RDNIS capabilites to route the caller to the correct mailbox and not the opening greeting. So with this would i still want to use the following to get the caller into my mailbox? dial-peer voice 2600 pots destination-pattern 2600 port 0/0/0:23 no digit-strip prefix 202555 ( assuming no LD code at this site ) this is the way i get callers into my mailbox - using RDNIS. On Wed, Mar 20, 2013 at 10:49 PM, William Bell wrote: If you are told National calls must present a 10D ANI AND you are restricted from using an alternate extension in CUC then I do the following. I am not sure whether this would be graded right or wrong On the SRST device (assume basic SRST) call-manager-fallback max-ephone 10 max-dn 20 oct huntst chan 1 voicemail 912025552699 ! or some unused DID on Site A call-forward noan 912025552600 time 20 !assume VM pilot is 2600 call-forward busy 912025552600 time-z time-f date-f call-forward pattern .T ! On CUCM: Create a PT: hq_gw-in_pt Create a CSS: hq_gw_css Assign CSS to hq gateway Either a.) create a translation in hq_gw-in_pt Pattern: 2699 xform ANI: xform DNIS: 2600! as in, redirect to regular VM pilot CSS: your regular HQ phone CSS will do OR b.) create a new hunt pilot in hq_gw-in_pt Pattern: 2699 HL: your VM HL xform ANI: Why would I go this path? 1. We had a requirement that National calls are presented with a 10D ANI in SRST mode. I assume that you would already have a translation-p that handles this bit 2. We can't modify the CUC subscriber. 3. This method doesn't interfere with RDNIS to VM 4. This method doesn't interfere with direct or redirect calls from HQ or SiteC Anyway, that is my 2 cents. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 20, 2013, at 9:33 PM, Bill wrote: Traditionally you would use the alternate extension or a on the pilot. So if you we're denied the ability to use alternate extension for this task but had to use it for another, say allowing easy voicemail access to a user at home, then I think you are looking at a very specific inbound translation on your gateway or nay sending 4 digits if the PSTN allows. I would definitely test out the translation setup to ensure you can do it. Sent from my iPad On Mar 20, 2013, at 3:44 PM, Steve Keller wrote: In SRST mode, when the vm button is pressed, i have a dial-peer to route this call to the vm hunt pilot on the UCM. dial-peer voice 2600 pots description voicemail-pilot destination-pattern 2600 no digit-strip port 0/0/0:23 prefix 1408202 If i have to adhere to the requirement that LD calls should be 10 digit ANI, then i am sending the full 10 digit ANI for this call as well ( even though it more of a hidden number rather than an implicit user dialed number) Thus the call arrives at site A GW with 10 digits , say 9723033002. In order to route this call to the correct mailbox i would have to use Alternate Extension of 9723033002 and then i will be prompted to login. However, if i am not allowed to use alternate extension then i must have another strategy. here are the choices i can think of, please chime in if you too have experienced this dilemma and what is the best way to solve it. 1) do not send the full 10 digit ANI for this call and it will arrive at site a GW as 4 digit ANI and then land in the mailbox, but not adhere to
Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension
As far as i remember the requirement under call routing doesn't specify the called party type "National" for long distance calls out SiteB gateway AND the question clearly said that YOU ARE ALLOWED TO SEND 4 DIGITS in Calling Number field to the Telco for these calls ... so why not use this dialpeer ..it still meets the requirement ... let me know if anyone has seen any different . dial-peer voice 2600 potsdescription voicemail-pilotdestination-pattern 2600no digit-stripport 0/0/0:23prefix 1408202 Mar 20 22:55:57.807: ISDN Se0/1/0:23 Q931: TX -> SETUP pd = 8 callref = 0x0084 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98397 Exclusive, Channel 23 Progress Ind i = 0x8183 - Origination address is non-ISDN Display i = '+19723033002' Calling Party Number i = 0x0080, '3002' Plan:Unknown, Type:Unknown Called Party Number i = 0x80, '14082022600' Plan:Unknown, Type:Unknown Date: Wed, 20 Mar 2013 17:01:48 -0500 From: wys...@gmail.com To: skeller...@gmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension Translation rule on CUCM. MGCP or H323 should not matter. You would match on the called number which is the voicemail pilot number then manipulate the calling number and send it on its way. It would not affect standard calls into Site A as it would not match the rule. Derek On Wed, Mar 20, 2013 at 4:51 PM, Steve Keller wrote: Thanks Derek On the surface it seems like that would chop down my ANI to 4 digits for any call into site A not just calls to vm. Also in my case site A is MGCP controlled so I that is not an option for me... On Mar 20, 2013 5:16 PM, "Derek Wyss" wrote: Alternatively, you could also create a translation rule in a partition accessible only by the inbound gateway that translates the calling number to 4 digits before sending it to voicemail. The hunt pilot calling transform mask will work, but you could have issues if you have any caller input requirements to route back out to the PSTN from UCON. Derek Wyss CCIE#38238(Voice) On Wed, Mar 20, 2013 at 3:44 PM, Steve Keller wrote: In SRST mode, when the vm button is pressed, i have a dial-peer to route this call to the vm hunt pilot on the UCM. dial-peer voice 2600 potsdescription voicemail-pilotdestination-pattern 2600 no digit-stripport 0/0/0:23prefix 1408202If i have to adhere to the requirement that LD calls should be 10 digit ANI, then i am sending the full 10 digit ANI for this call as well ( even though it more of a hidden number rather than an implicit user dialed number) Thus the call arrives at site A GW with 10 digits , say 9723033002.In order to route this call to the correct mailbox i would have to use Alternate Extension of 9723033002 and then i will be prompted to login. However, if i am not allowed to use alternate extension then i must have another strategy. here are the choices i can think of, please chime in if you too have experienced this dilemma and what is the best way to solve it. 1) do not send the full 10 digit ANI for this call and it will arrive at site a GW as 4 digit ANI and then land in the mailbox, but not adhere to LD calls should be 10 digit ANI requirement. 2) put as calling party transform mask on the Hunt Pilot, thus stripping the caller ANI to 4 digits and i can be prompted to log in. However i think with this method, anytime the caller ANI is read to before the message is played the caller id would incorrectly state from "3002" instead of from "9723033002" essentially, what is the best way for SRST users to access voicemail when you are not permitted to use Alternate Extension. thanks in advance all!! steve ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] H323 VG dial-peers
Based on my guesstimate these are the mandatory commands needed to make the script happy. dial-p v 1 voipdesc SUB-dialpeervoice-class codec 1voice-class h323 1no vaddestination-pattern 3...$session target ipv4:10.10.210.11 -->SUBip qos dscp cs3 signalpreference 0dtmf-relay h245-alpha!!!dial-p v 2 voipdesc PUB-dialpeervoice-class codec 1voice-class h323 1no vaddestination-pattern 3...$session target ipv4:10.10.210.10 -->PUBip qos dscp cs3 signalpreference 1dtmf-relay h245-alpha! Setup and TCP timeout should be set to 3 seconds each under the h323 class. Date: Sun, 10 Mar 2013 00:36:16 +0300 From: aboaz...@gmail.com To: amccar...@cciequest.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] H323 VG dial-peers Hi, Thanks all, Got it On Sun, Mar 10, 2013 at 12:01 AM, Amp wrote: Hey, I would hard code the preferences. Being that the default preference in canonical terms is 0, you could set the preference on the pub dial peer to preference 1, or you could set the sub dial-peer to preference 1 and the pub to preference 2. Either way, I would hard code it. Amp Quoting CCIEing : I am sorry the below dial-peers should be : dial-peer voice 2 voip destination-pattern 12341$ session target ipv4:Sub_IP_Address codec g711ulaw dtmf-relay h245-alphanumaric h245-siganal cisco no vad dial-peer voice 3 voip destination-pattern 12341$ session target ipv4:Pub_IP_Address codec g711ulaw dtmf-relay h245-alphanumaric h245-siganal cisco no vad On Sat, Mar 9, 2013 at 11:14 PM, CCIEing wrote: Hi All, For site B let us say that the lab ask us to configure it as H323 gateway, and the question mentioned that call have to go to SUB CUCM then to PUB, in that case we have to create the 2 voip dial peers pointing to sub and PUB as below : My question here , in order to make sure the preference of the 1st dial peer , do we have to hard coded the preference command in the dial-peers or the tag index of the dial peer will grantee that the sup dial peer will chosen 1st as it is tag is less than the pub dial-peer Thanks dial-peer voice 2 voip destination-pattern 12341$ session target ipv4:Sub_IP_Address codec g711ulaw dtmf-relay h245-alphanumaric h245-siganal cisco no vad dial-peer voice 2 voip destination-pattern 12341$ session target ipv4:Sub_IP_Address codec g711ulaw dtmf-relay h245-alphanumaric h245-siganal cisco 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] B Channel Busy Out
The B-channel selection order is always default i.e, descending. Date: Sat, 9 Mar 2013 22:29:01 +0300 From: aboaz...@gmail.com To: jas7...@gmail.com CC: ccie_voice@onlinestudylist.com; garyclark...@gmail.com Subject: Re: [OSL | CCIE_Voice] B Channel Busy Out Hi Jason, I have question about step # 9 clear counters is it the normal clear counter command? e.x: E1 cardclear counter interface s x/y/z:15 Thanks On Wed, Feb 27, 2013 at 3:02 PM, Jason Lee wrote: I use this as a strategy for checking my gateway configuration Ensure that your are meeting requirements on the following display-ieBCHAN order selection (Ascending, Descending)BCHAN numberHow many BCHANs? If not specified create a full PRI. If fractionalSet BCHAN Maintenance in Advanced Service ParametersCheck the Check Status checkbox in GW configClockingNetwork clock participate network clock select 1 t10/0/0ISDN Switch-TypeSource-Address911Done in gateway section. Make sure to have routed correctly, SLRG?, Direct Inward Dial clear counters On Wed, Feb 27, 2013 at 3:38 AM, Jamie Parr (jamparr) wrote: I am also curious as to the grading on the gateways, I received very low marks on this section. Can anyone help? Thanks Jamie Parr Engineer - IT jamp...@cisco.com Phone: +44 20 8824 2641 Mobile: +44 7590622049 From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Pixar Perfect Sent: 26 February 2013 19:35 To: Steve Keller; GARY CLARK Cc: CCIE Voice OSL Subject: [OSL | CCIE_Voice] B Channel Busy Out Gary ..you mentioned B channel busyout on service parameter. in my understanding this was only needed when you would "download" the GW config from CCM i.e., ccm-manager config. it doesn't make any sense to use this service parameter as most of the solution guides (INE, IPXEPERT, 360) do not encourage the use of ccm-manager config except initial stage of your config and then disable it. I have heard ppl who passed just using standard configs but not sure if they did the B channel busy out on service parameter. mgcp mgcp call-agent 10.10.210.11 -->sub mgcp dtmf mgcp bind ... (2x2) ccm-mana fall ccm-mana music ccm-mana mgcp ccm-mana red 10.10.210.10 --> pub if B channel status is really graded on the exam then it is one of those things that doesn't make sense to have it there but is needed to score points experts, any comments or advise from the recent Experts ? PIXAR Date: Mon, 25 Feb 2013 14:31:12 -0500 From: skeller...@gmail.com To: garyclark...@gmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] lab>7 failed for 1% I recieved 29% in RTP on GW Signalling section and Call Routing as well. I am very discouraged i could score very low marks on these sections as i took my time and felt like i had nailed them. I scored really well in all other areas but failed because of these 2 sections. It is a mystery to me what the proctor is doing to arrive at that score, when all my calls worked, the debugs matched the requirements, i was binding to the correct interfaces, setting up the correct protocol and channels,etc. I would love to hear what insight folks have as to why the scores could be so low when everything looked to be working beautifully, without breaking NDA of course. thanks steve On Mon, Feb 25, 2013 at 1:54 PM, GARY CLARK wrote: Hi friends, I got lab 7 in SJC and i completed that in 6 hrs and tested for 2 hrs time. I thought i have passed 1000% but when i saw my result i was surprised. I almost got everywhere 100% except VG / 29% which was 17 marks section. Same story with my friends do anyone got 100% in VG for lab 7 If anyone interested to share the hidden secrets then welcome as people are getting lab 7 repeating now very eager to understand what could be wrong. Please email me for further discussion. We 3 friends attempted out of which i also did busy out channel but that also did not helped its 29% only why so Regards ___ 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 v
Re: [OSL | CCIE_Voice] Phone NTP Reference Vs NTP Reference
you still need the Phone NTP reference on the labs as CUPC client is a SIP client ..there are no SIP phones on the Version 3 labs but we might see lot on Version 4. Date: Tue, 5 Mar 2013 01:05:22 +0300 From: aboaz...@gmail.com To: corygray22...@hotmail.com; bring...@gmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Phone NTP Reference Vs NTP Reference Oh thanks a lot for your input. Appreciated .. On Tue, Mar 5, 2013 at 12:48 AM, Cory Gray wrote: Phone ntp reference is for SIP phones only Sent from my iPhone On Mar 4, 2013, at 4:42 PM, "CCIEing" wrote: > Hello All, > > The following question cross my mind while doing the NTP configuration stuff.. > > What is the difference between the Phone NTP reference configuration in the > CCM Web administration page > and > The NTP reference on the OS Administration page?? > > does the 1st one for the endpoints where the 2nd one is for the CUCM itself? > > 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
[OSL | CCIE_Voice] B Channel Busy Out
Gary ..you mentioned B channel busyout on service parameter. in my understanding this was only needed when you would "download" the GW config from CCM i.e., ccm-manager config. it doesn't make any sense to use this service parameter as most of the solution guides (INE, IPXEPERT, 360) do not encourage the use of ccm-manager config except initial stage of your config and then disable it. I have heard ppl who passed just using standard configs but not sure if they did the B channel busy out on service parameter. mgcp mgcp call-agent 10.10.210.11 -->submgcp dtmf mgcp bind ... (2x2) ccm-mana fallccm-mana musicccm-mana mgcpccm-mana red 10.10.210.10 --> pub if B channel status is really graded on the exam then it is one of those things that doesn't make sense to have it there but is needed to score points 🙉 experts,any comments or advise from the recent Experts ? PIXAR Date: Mon, 25 Feb 2013 14:31:12 -0500 From: skeller...@gmail.com To: garyclark...@gmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] lab>7 failed for 1% I recieved 29% in RTP on GW Signalling section and Call Routing as well. I am very discouraged i could score very low marks on these sections as i took my time and felt like i had nailed them. I scored really well in all other areas but failed because of these 2 sections. It is a mystery to me what the proctor is doing to arrive at that score, when all my calls worked, the debugs matched the requirements, i was binding to the correct interfaces, setting up the correct protocol and channels,etc. I would love to hear what insight folks have as to why the scores could be so low when everything looked to be working beautifully, without breaking NDA of course. thankssteve On Mon, Feb 25, 2013 at 1:54 PM, GARY CLARK wrote: Hi friends, I got lab 7 in SJC and i completed that in 6 hrs and tested for 2 hrs time. I thought i have passed 1000% but when i saw my result i was surprised. I almost got everywhere 100% except VG / 29% which was 17 marks section. Same story with my friends do anyone got 100% in VG for lab 7 If anyone interested to share the hidden secrets then welcome as people are getting lab 7 repeating now very eager to understand what could be wrong. Please email me for further discussion. We 3 friends attempted out of which i also did busy out channel but that also did not helped its 29% only why so Regards ___ 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
[OSL | CCIE_Voice] SCCP CCM GROUP config
hello, any need for these commands? i saw these on IPExpert configurations and INE guides in some of their labs. sccp ccm gr 1ass ccm 1 pr 1ass ccm 2 pr 2ass ccm 3 pr 3 --> conf registration for srstass pro 1 reg br2cfbass pro 2 reg br2transwitchback method immediateswitchover method immediatesccp interface Vlan112 --> voice vlan ( i think this makes sense)keepalive retries 2keepalive timeout 10registration timeout 10registration retry 2audio dscp efsignaling dscp cs3 these might be needed to register conference resource faster on the SRST GW but do we need to use these in the exam?? any "Best practice" advice. thanks. keepalive retries 2keepalive timeout 10registration timeout 10registration retry 2___ 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] RSVP MTP Codec passthru
hello experts, reading thru sample RSVP configs on the OSL forum i see the wide of use codec passthru under software mtp configuration for rsvp , any reason why codec passthru needed on the mtp. i have it working always without this command. or is this again one of those configuration pieces that needs to be there but no real reason :) many 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
[OSL | CCIE_Voice] Recording AU file
What is the quickest way to record AU file in the lab for BACD file needs? i tried recording script on the IPCC Express but it dumps WAV file. CUC recording applet never works on my windows box. any better way? 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
[OSL | CCIE_Voice] MWI Best Practice
Experts and wannabe experts friends, what are the best practices for MWI in CME and SRST modes for the CUE site BR2? i was used to using MWI ON and MWI OFF DNs on a CME but i was told by a fellow aspirant that MWI ON/OFF are not preferred (grading wise) and that solicited MWI is that gets you to the needed points. however i have seen solicited and unsolicited to be verify unreliable on 7965 phones .. you have to do no mwi sip and mwi sip to get solicited to work and sometimes reboot CUE or router to get both solicited and unsolicited to work. I am 1 month away from exam date and dont want to waste time exploring instead adopt best common practice that works flawlessly ..and so far it has been ON/OFF DN ___ 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] cBarge & Barge softkey
When working on Shared DNs and cBarge question (5-lab handbook, Lab1) that needs use of CFB on Site C, do we need to remove the Barge Softkey from the Remote in Use state? do you think it is good idea to disable Built in Bridge for the two phones that have a shared line and need GW CFB for conferencing.? the solution guide has an example that has the Barge softkey left there in Remote In Use. Per IPEXPERT's bootcamp, the recommendation was not to tamper with the existing Softkey layout and keep adding softkeys. It makes sense however, this particular Barge vs cBarge is tricky thing ... i would be least worried abt these things but it will be unfortunate if the script is looking for Barge softkey as well :) ... the notorious grading script process worries me as it is the deal breaker :) thx...pixar ___ 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 transfer system and pattern
Thanks, makes sense. One of those few configurations on the exam that sticks to the design guidelines & field deployments. :) :) Date: Sat, 16 Feb 2013 17:48:16 -0600 Subject: Re: [OSL | CCIE_Voice] SRST transfer system and pattern From: ramcharan.a...@gmail.com To: corygray22...@hotmail.com CC: pixarperf...@live.com; ccie_voice@onlinestudylist.com Hi, As per cisco CME design guide these commands are necessary. Please refer cisco CME SRND. http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/srnd/design/guide/clproc.html#wp1068396 Regards, Ramcharan Arya CCIE # 28926 ( R&S) On Fri, Feb 15, 2013 at 4:51 PM, Cory Gray wrote: I have had several conversations with people on this. Everyone can easily make SRST work but scoring points seems to be the trickiest thing in the lab. So I do not think anyone knows for sure what should or should not be on the “template” I have never scored any points there so I cannot give an OPINION on what should or should not be there. People say they score points and then go with the same template on the next lab and get 0 so it is a mystery. People can share templates without breaking NDA since the question is never discussed. Getting the question right is the easy part! From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Pixar Perfect Sent: Friday, February 15, 2013 5:26 PM To: CCIE Voice OSL Subject: [OSL | CCIE_Voice] SRST transfer system and pattern transfer-system full-consultdo we need to specify this? I thought by default it is wnabled but I read on voiceie forum someone scored nothing on SRST adn the only conclusion was the transfersystem consult was missing. Any thoughts? srst mode auto-provision all srst ephone description SRST-EPHONES-CME srst dn template 1 srst dn line-mode octo max-ephones 10 max-dn 10 preference 2 no-reg both ip source-address 10.10.1.13 port 2000 time-zone 42 max-conferences 8 gain -6 call-forward pattern .T time-webedit transfer-system full-consult transfer-pattern .T ___ 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] SRST transfer system and pattern
transfer-system full-consultdo we need to specify this? I thought by default it is wnabled but I read on voiceie forum someone scored nothing on SRST adn the only conclusion was the transfersystem consult was missing. Any thoughts? srst mode auto-provision all srst ephone description SRST-EPHONES-CME srst dn template 1 srst dn line-mode octo max-ephones 10 max-dn 10 preference 2 no-reg both ip source-address 10.10.1.13 port 2000 time-zone 42 max-conferences 8 gain -6 call-forward pattern .T time-webedit transfer-system full-consult transfer-pattern .T ___ 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] ISDN signaling config
Is there a need to enable(check) "Setup non-ISDN Progress Indicators IE Enable" on the MGCP GW page ?___ 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] Messages button in SRST
thx bill will try it out before the end of this week and let u know. Date: Tue, 5 Feb 2013 06:15:35 -0600 Subject: Re: [OSL | CCIE_Voice] Messages button in SRST From: whl...@gmail.com To: pixarperf...@live.com CC: ccie_voice@onlinestudylist.com Why not just set up calling party transformations that strip the extra digits and apply it to the vm hunt. Make the patterns only match the incoming numbers from site B? In IPExpert advanced labs (OWLE/5 Lab pack) that would be 408387300X so just make the pattern 408387.300X and strip it to the dot On Tue, Feb 5, 2013 at 2:35 AM, Pixar Perfect wrote: When Site B Phone 2 (DN 3002) hits the messages button in SRST, the SRST call control dials 912025552220 to the PSTN to ring the Voicemail ports at Site A. Unity plays generic Welcome greeting instead of the user greeting of user 3002. The requirement is NOT to use Alternate Extension on the user. Another requirement on the Voicemail question is customize user 3002 so user can hear the ANI and timestamp of the caller. IPExperts solution to the 1st requirement is to use calling number mask under hunt pilot. However this breaks the second requirement, as Unity Connection will only receive 4 digits ANI even for standard PSTN callers. Any comments of this use of could be graded and any experiences? I have a workaround that would cater to users at SiteB hitting messages button to check messages: 1) Setup the voicemail DN in SRST to 912025552225 2) User hits messages button and SRST/CME dials 912025552225 to Site A 3) At the CUCM, setup a translation pattern to translate 2225/pt-gw-sa-only to 2220 and this same translation pattern masks the calling number . CSS (css-internal) on the translation pattern would be able to access the VM hunt pilot 2220 4) Site B dialing 2225 for VM access would give the CUCM intelligence to differentiate a PSTN call to VM versus a call from Site B to VM via PSTN. Due to the translation pattern configured the calling number presented to the CUC would be i.e., 3002 CFNA and CFB at Site B SRST would still use 912025552220. Here is the call flow for CFNA and CFB in SRST 1) PSTN phone calls Site B Phone 2 (SB-Ph2) while Site B is in SRST 2) SB-Ph2 ring no-answer to voicemail DN which is setup as 912025552220 3) Call hairpins back to PSTN to dial the VM pilot 2220 at Site A. ISDN setup carries RDNIS set to 3002 4) CUCM receives RDNIS in the call, rings the hunt pilot 2220 and delivers it to Unity Connection to identify VM Box. I have tested these approaches in the lab and each works like a charm, not sure if proctor deems this as a valid solution. Any thoughts? ___ 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] Messages button in SRST
When Site B Phone 2 (DN 3002) hits the messages button in SRST, the SRST call control dials 912025552220 to the PSTN to ring the Voicemail ports at Site A. Unity plays generic Welcome greeting instead of the user greeting of user 3002. The requirement is NOT to use Alternate Extension on the user. Another requirement on the Voicemail question is customize user 3002 so user can hear the ANI and timestamp of the caller. IPExperts solution to the 1st requirement is to use calling number mask under hunt pilot. However this breaks the second requirement, as Unity Connection will only receive 4 digits ANI even for standard PSTN callers. Any comments of this use of could be graded and any experiences? I have a workaround that would cater to users at SiteB hitting messages button to check messages: 1) Setup the voicemail DN in SRST to 912025552225 2) User hits messages button and SRST/CME dials 912025552225 to Site A 3) At the CUCM, setup a translation pattern to translate 2225/pt-gw-sa-only to 2220 and this same translation pattern masks the calling number . CSS (css-internal) on the translation pattern would be able to access the VM hunt pilot 2220 4) Site B dialing 2225 for VM access would give the CUCM intelligence to differentiate a PSTN call to VM versus a call from Site B to VM via PSTN. Due to the translation pattern configured the calling number presented to the CUC would be i.e., 3002 CFNA and CFB at Site B SRST would still use 912025552220. Here is the call flow for CFNA and CFB in SRST 1) PSTN phone calls Site B Phone 2 (SB-Ph2) while Site B is in SRST 2) SB-Ph2 ring no-answer to voicemail DN which is setup as 912025552220 3) Call hairpins back to PSTN to dial the VM pilot 2220 at Site A. ISDN setup carries RDNIS set to 3002 4) CUCM receives RDNIS in the call, rings the hunt pilot 2220 and delivers it to Unity Connection to identify VM Box. I have tested these approaches in the lab and each works like a charm, not sure if proctor deems this as a valid solution. Any thoughts? ___ 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] CUCM to Gatekeeper H323-GW vs. VOIP-GW
CUCM registers to the GK as a VOIP-GW. How do I make it register as a H323-GW? The requirement on a practice lab demands the CUCM register to the GK as a H323-GW and not VOIP-GW. On a GW,, you could just do the allow h323 to h323 to make it register as H323-GW. But not sure abt CUCM. Any thought? ___ 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] Connection Monitor Duration for SRST testing
Hello, is it advisable to bring down the Connection Monitor Duration under Device Pool to a low value like 30 seconds to expedite the SRST testing in the lab? Does it affect grading if we happen to put it back to default? thx ___ 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] CUC PhoneSystem configurations
On the CUC, here are the following checkboxes I typically enable (check). Any other configuration under the phone system? Default TRAP Switch Enable for Forwarded Message Notification Calls (by Using DTMF) Enable for Forwarded Message Notification Calls (by Using Extension) Send Message Counts Any suggestions? thx ___ 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 Best Pract
Ignore the RTP header compressions part. My question is not technical but from strategy point. Do you think the grading could be affected if we remove the names built by the QOS script and use our custom names? Thanks Date: Sat, 5 Jan 2013 08:55:57 +0100 Subject: Re: [OSL | CCIE_Voice] QOS Best Pract From: stbruen...@gmail.com To: pixarperf...@live.com CC: ccie_voice@onlinestudylist.com The problem with your example is that the frame-relay ip rtp header-compression is out-dated and the header compression should be in the policy-map in the rtp class. Also I don't see any less flexibilty in using auto qos, you can adjust all the values as with the manual way. Regards Steffen Am Samstag, 5. Januar 2013 schrieb Pixar Perfect : What is the best practice for the real lab QOS after running the AutoQos - use the AutoQoS nomenclature and make changes to the classmaps, policymaps and map-class OR copy the AutoQOS output on a notepad and rename policymaps, classmaps and mapclass? I typically follow the second route as it give me more flexibility to "finalize" the configs on the notepad and then just copy paste instead of changing the AutoQOS configs on the router itself. Any inputs , comments or suggestions especially from those who had success with the second approach? Example: !class-map match-any RTP match ip dscp ef class-map match-any CONTROL match ip dscp cs3 ! policy-map VOIP class RTPpriority 24 class CONTROLbandwidth 19 class class-defaultfair-queue ! !interface Serial0/1/1:0.1 point-to-point bandwidth 384 frame-relay interface-dlci 201 class FRVOIP frame-relay ip rtp header-compression ! !!map-class frame-relay FRVOIP frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 364800 frame-relay fragment 480 service-policy output VOIP ! ___ 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] QOS Best Pract
What is the best practice for the real lab QOS after running the AutoQos - use the AutoQoS nomenclature and make changes to the classmaps, policymaps and map-class OR copy the AutoQOS output on a notepad and rename policymaps, classmaps and mapclass? I typically follow the second route as it give me more flexibility to "finalize" the configs on the notepad and then just copy paste instead of changing the AutoQOS configs on the router itself. Any inputs , comments or suggestions especially from those who had success with the second approach? Example:!class-map match-any RTP match ip dscp ef class-map match-any CONTROL match ip dscp cs3 !policy-map VOIP class RTPpriority 24 class CONTROL bandwidth 19 class class-defaultfair-queue! !interface Serial0/1/1:0.1 point-to-point bandwidth 384 frame-relay interface-dlci 201 class FRVOIP frame-relay ip rtp header-compression! !!map-class frame-relay FRVOIP frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 364800 frame-relay fragment 480 service-policy output VOIP! ___ 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] UCCX one button login
in my experience the phone had to be factory reset. make sure there is a valid DHCP that phone can access after the factory reset sequence. To: ch.christ...@logicom.net; b...@ucguerrilla.com From: singh8...@in.com Date: Sat, 5 Jan 2013 10:00:46 +0530 CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] UCCX one button login Hi Christof, I have checked it and it is in the correct order. I rebooted the UCCX server and tried login via one button Now I can see the agent in not ready state . If I try getting the agent into ready state . I see the following message... "Cannot change agent state because phone is out of service" Anything else I can check? -singh -- Original message -- From:"Chrysostomos Christofi"< ch.christ...@logicom.net > Date: 2 Jan 13 22:26:55 Subject: RE: [OSL | CCIE_Voice] UCCX one button login To: ; "b...@ucguerrilla.com" Cc: "ccie_voice@onlinestudylist.com" Hi Did you configure exactly as the below the parameters?(case sensitive) 1) Ext 2) ID 3) Pwd From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of singh Sent: Τετάρτη, 2 Ιανουαρίου 2013 6:51 μμ To: b...@ucguerrilla.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] UCCX one button login hi Guys, Thanks for your inputs. I changed the url to agentLogin.jsp . Now when I press the services button and then go to IPPA . I see the following error message... "Either the agent ID or password u entered is invalid" -I have double checked the agent ID and pwd and it is correct -I have added the phones to the RM user. -I have a resource group added to the IPCC extensions on UCCX What else do I do? -singh -- Original message -- From:"William Bell"< b...@ucguerrilla.com > Date: 2 Jan 13 00:00:03 Subject: Re: [OSL | CCIE_Voice] UCCX one button login To: Cc: whl...@gmail.com, ccie_voice@onlinestudylist.com Which URL are you using for the phone service. Sounds like you are using: http://:6293/ipphone/jsp/sciphonexml/IPAgentInitial.jsp ...as opposed to the following URL: http://:6293/ipphone/jsp/sciphonexml/IPAgentLogin.jsp The first URL will provide an input menu to the IP phone in a manner identical to what you describe. The latter URL, when provisioned with the correct parameters, will get you the one button login experience you are looking for. HTH -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Jan 1, 2013, at 9:05 AM, singh wrote: hi Bill, Those fields are added to the users under the subscricption for each phone. However when I press the services button on the phone it still prompts me for the... Name Password Extension Instead of directly logging into the service. -- Original message -- From:"Bill"< whl...@gmail.com > Date: 31 Dec 12 23:28:24 Subject: Re: [OSL | CCIE_Voice] UCCX one button login To: Cc: ccie_voice@onlinestudylist.com So when you assign the service to the phone add those fields for each user, also when you create the service you can set defaults so one user works right away and you just edit the rest Bill On Dec 31, 2012, at 5:19 AM, "singh" wrote: hi Guys, I have configured 1 button login for my agent phones however when I press the services button it takes me to name , ext and password . I would like to configure it so that it directly logs in after I press the services button instead of having to enter the name , password and id -singh Get Yourself a cool, short @in.com Email ID now! ___ 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 Get Yourself a cool, short @in.com Email ID now! ___ 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 Get Yourself a cool, short @in.com Email ID now! Get Yourself a cool, short @in.com Email ID now! ___ 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] 'Phone Services Debate' in CCIE_Voice digest Vol 81, Issue 35
Rob ..ditto. i agree that the method is little time consuming but provided you have the URL and XML syntax without errors (it is just copy paste), it sounds more acceptable solution to meet the requirements on the question from the lab experience lab 1. Date: Sat, 17 Nov 2012 18:58:04 + From: gri...@ymail.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] 'Phone Services Debate' in CCIE_Voice digest Vol 81, Issue 35 In reply to the questions regarding subscriptions to Phone Services. I have been playing with this over the last few days. I have come to the conclusion that the best method is to upload .xml files to the UCCX server. This is achieved by: RDP to UCCX. Navigate to c:\Inetput\wwwrootCreate a new text file and name it default.xmlAdd the relevant services to this file*Delete the services in 'Phone Services' to prevent duplicate entries in the phone's menuNavigate to Enterprise subscriptions and make these changes:"Services Provisioning" = "both""URL Directories" = http://ip_of_UCCX/default.xmlIf an individual phone requires a custom menu:Create a custom .xml file and host it on UCCX - e.g. hqph1.xmlNavigate to the phone (e.g. HQ PH1) in CUCM and change "Directory" to http://ip_of_UCCX/hqph1.xmlThis will allow you to assign custom menus to phones and also have a custom template for all phones in the environment. It also preserves the use of the Voicemail button, which can be disabled by some other methods. Regards, Rob * a template for the .xml file can be found in the CUCM programming guide. From: "ccie_voice-requ...@onlinestudylist.com" To: ccie_voice@onlinestudylist.com Sent: Saturday, 17 November 2012, 16:29 Subject: CCIE_Voice Digest, Vol 81, Issue 35 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: No Services Configured (Mohamed Gazzaz) 2. Re: No Services Configured (Edgar Feliz) -- Message: 1 Date: Sat, 17 Nov 2012 19:13:36 +0300 From: Mohamed Gazzaz To: Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] No Services Configured Message-ID: Content-Type: text/plain; charset="windows-1256" Thanks for the clarification. Can't we achieve the same result by using these steps - Delete the Missed, Placed, Corporate, Placed and Received from the Phones > IP Phone Services page - From the below link, restore the 5 services with "enterprise subscription" --> f (false) http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/rel_notes/7_0_1/cucm-rel_notes-701.html#wp631159 - Use BAT to subscribe the 5 services to the desired phones I tried this and it worked for me. Regards, Mohamed Gazzaz Subject: Re: [OSL | CCIE_Voice] No Services Configured From: w...@netcraftsmen.net Date: Sat, 17 Nov 2012 11:01:46 -0500 CC: ke...@kevinspicer.co.uk; pixarperf...@live.com; ccie_voice@onlinestudylist.com To: mgaz...@hotmail.com The OP was referencing one of the four labs in the "One Week Lab Experience (OWLE)" workbook. It is a different set of labs than the 5-Day Lab workbook. -Bill --William Bellblog: http://ucguerrilla.comtwitter: @ucguerrilla On Nov 17, 2012, at 5:29 AM, Mohamed Gazzaz wrote:Where is this solution ? which lab and question ? I searched for it but could not find it in the 5 labs self study workbook. Date: Sat, 17 Nov 2012 00:28:52 + From: ke...@kevinspicer.co.uk To: pixarperf...@live.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] No Services Configured The enterprise url parameters apply only to phones using external provisioning. So the IP expert solution is fine. Be aware that some older phone models (e.g.7960) only support external provisioning so you might see different results depending on what hardware you have - but the Cisco lab itself is 7965 phones (this is public published info) On 17 Nov 2012 00:01, "Pixar Perfect" wrote: I am working on the question from the lab experience handbook where SA phone 1 needs to see "No Services Configured" when it hits directory button. IPEXPERT suggests that we delete the Service URL from EP and then setup the SA Phone 1 device configuration page Service parameter to External ? I dont think it is acceptable solution as this could cause other phones to not see any services if they reboot and d
Re: [OSL | CCIE_Voice] No Services Configured
that is one limitation of just deleting it and then readding it as a regular directory service without enterprise subscription. So far I have found the UCCX solution seamless. Date: Sat, 17 Nov 2012 11:46:41 -0600 From: whl...@gmail.com To: mgaz...@hotmail.com CC: ccie_voice@onlinestudylist.com; w...@netcraftsmen.net Subject: Re: [OSL | CCIE_Voice] No Services Configured Did it place them in the correct order on your phone when you press services? I did this manually and they are not in the same order without being enterprise services. On Sat, Nov 17, 2012 at 10:13 AM, Mohamed Gazzaz wrote: Thanks for the clarification. Can't we achieve the same result by using these steps - Delete the Missed, Placed, Corporate, Placed and Received from the Phones > IP Phone Services page - From the below link, restore the 5 services with "enterprise subscription" --> f (false) http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/rel_notes/7_0_1/cucm-rel_notes-701.html#wp631159 - Use BAT to subscribe the 5 services to the desired phones I tried this and it worked for me. Regards, Mohamed Gazzaz Subject: Re: [OSL | CCIE_Voice] No Services Configured From: w...@netcraftsmen.net Date: Sat, 17 Nov 2012 11:01:46 -0500 CC: ke...@kevinspicer.co.uk; pixarperf...@live.com; ccie_voice@onlinestudylist.com To: mgaz...@hotmail.com The OP was referencing one of the four labs in the "One Week Lab Experience (OWLE)" workbook. It is a different set of labs than the 5-Day Lab workbook. -Bill --William Bellblog: http://ucguerrilla.comtwitter: @ucguerrilla On Nov 17, 2012, at 5:29 AM, Mohamed Gazzaz wrote: Where is this solution ? which lab and question ? I searched for it but could not find it in the 5 labs self study workbook. Date: Sat, 17 Nov 2012 00:28:52 + From: ke...@kevinspicer.co.uk To: pixarperf...@live.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] No Services Configured The enterprise url parameters apply only to phones using external provisioning. So the IP expert solution is fine. Be aware that some older phone models (e.g.7960) only support external provisioning so you might see different results depending on what hardware you have - but the Cisco lab itself is 7965 phones (this is public published info) On 17 Nov 2012 00:01, "Pixar Perfect" wrote: I am working on the question from the lab experience handbook where SA phone 1 needs to see "No Services Configured" when it hits directory button. IPEXPERT suggests that we delete the Service URL from EP and then setup the SA Phone 1 device configuration page Service parameter to External ? I dont think it is acceptable solution as this could cause other phones to not see any services if they reboot and download new config file which has no directories on any phone !!! My approach as follows1) make a directory.xml file using the IP Phone Services document from cisco.com2) delete the Missed, Placed, Corporate, Placed and Received from the Phones > IP Phone Services page 3) Setup the EP for Services to Both & Update the URL to http://x.x.x.x/directory.xml where x.x.x.x is UCCX IP4) Upload the directory.xml to the IIS server on the UCCX, restart IIS 5) on SA Phone 1 page, setup the Services to Internal, Reset all phones Comments? Feedback? Experiences? TIA. directory.xml Missed Calls Application:Cisco/MissedCalls CorporateDirectory Application:Cisco/CorporateDirectory Received CallsApplication:Cisco/ReceivedCalls Placed Calls Application:Cisco/PlacedCalls Personal Directory Application:Cisco/PersonalDirectory ___ 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 __
[OSL | CCIE_Voice] No Services Configured
I am working on the question from the lab experience handbook where SA phone 1 needs to see "No Services Configured" when it hits directory button. IPEXPERT suggests that we delete the Service URL from EP and then setup the SA Phone 1 device configuration page Service parameter to External ? I dont think it is acceptable solution as this could cause other phones to not see any services if they reboot and download new config file which has no directories on any phone !!! My approach as follows1) make a directory.xml file using the IP Phone Services document from cisco.com2) delete the Missed, Placed, Corporate, Placed and Received from the Phones > IP Phone Services page3) Setup the EP for Services to Both & Update the URL to http://x.x.x.x/directory.xml where x.x.x.x is UCCX IP4) Upload the directory.xml to the IIS server on the UCCX, restart IIS5) on SA Phone 1 page, setup the Services to Internal, Reset all phones Comments? Feedback? Experiences? TIA. directory.xml Missed Calls Application:Cisco/MissedCalls CorporateDirectory Application:Cisco/CorporateDirectory Received CallsApplication:Cisco/ReceivedCalls Placed Calls Application:Cisco/PlacedCalls Personal Directory Application:Cisco/PersonalDirectory ___ 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] CS to DSCP?
Guys any suggestions as to which is the best way to remember the class selectors to DSCP mappings? I mean CS3 to 24 and CS1 to 8 ... any easy trick? 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
[OSL | CCIE_Voice] LAN Qos questions
The requirement is as follows on Lab 2 QOS section of the 5-Lab Handbook. "For traffic being sent to the Site A gateway ensure that the traffic marked with COS 5 is dropped if the queue 1 is 75% full" The Solution guide (page 408) has the following solution. mls qos queue-set output 2 threshold 1 75 100 100 100 --> queset is preconfigured on the port to 2mls qos srr-queue output cos-map queue 1 threshold 3 5 ..My interpretation was to move the Cos 5 into Q1t1 but the command says threshold 3 .. is this just a typo or am I missing something obvious. 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
