Re: [OSL | CCIE_Voice] CCIE #26721 - I PASSED!
Congrats I got my number on Monday, so the two Matts are in!! On Aug 17, 2010, at 5:05 PM, Matthew Berry wrote: I just got my score report. I passed guys. More follow-up to come later. Right now I'm now on cloud nine. :) CCIE #26271 Thanks, Matthew Berry ciscovoiceg...@gmail.com http://ciscovoiceguru.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Gatekeeper IPIPGW
outvia sends the call to the zone specified on the way to the destination zone (like a next hop statement in a policy routebut not really). The gatekeeper then just looks for a cube registered in that outvia zone. The cube has to have a matching inbound voip dialpeer AND a outgoing dialpeer with a destination pattern that matches the outbound digits and session target RAS. The CUBE uses that dial-peer to send the call back to the gatekeeper, the gatekeeper knows then to send the call on to the final zone without looping back to the CUBE. Matt On Aug 12, 2010, at 1:19 PM, CCIE Voice GMAIL wrote: Hi Matt, I wanted to see what the logic was behind the zone local statement. Is it good practice to do it that way (invia and outvia) for Intra-zone GK routing? From what I would understand, this is almost implied by just saying zone local GK Can you share with me your reasoning for doing it this way? Thanks in advance. -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccieid1ot Sent: Thursday, August 12, 2010 8:40 AM To: Matthew Berry Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Gatekeeper IPIPGW Use this on your local GK. gatekeeper zone local GK cisco.com x.x.x.x invia GK outvia GK enable-intrazone zone remote BBGK cisco.com x.x.x.x1719 zone prefix BBGK 01132* zone prefix BBGK 01144* no shutdown On Wed, Aug 11, 2010 at 9:01 PM, Matthew Berry ciscovoiceg...@gmail.com wrote: Edwin - You need to add the outvia command to the end of your remote zone and specify the zone that has your IPIPGW. Thanks, Matthew Berry ciscovoiceg...@gmail.com http://ciscovoiceguru.com On Aug 11, 2010, at 7:13 PM, Edwin Dotson wrote: Sorry for all the same topic. I have successfully got my 2 gatekeepers talking and calls back and fourth. But now I can’t get calls to pass through the IPIPGW it sends the calls directly to the Remote Gatekeeper. To identify calls destined for the IPIPGW I have been using 3#. LOCAL GK/IPIPGW version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname LOCAL ! boot-start-marker boot system flash flash:c3825-adventerprisek9_ivs-mz.124-25c.bin boot-end-marker ! enable password ! no aaa new-model ip cef ! ! ! ! ip auth-proxy max-nodata-conns 3 ip admission max-nodata-conns 3 ! voice-card 0 no dspfarm ! ! ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip supplementary-service h450.12 ! ! ! ! ! ! ! ! ! ! voice translation-rule 1 rule 1 /^3#/ // ! ! voice translation-profile strip3# translate called 1 ! ! ! ! ! ! ! ! ! ! ! interface GigabitEthernet0/0 ip address 10.201.3.30 255.255.255.0 duplex auto speed auto media-type rj45 ! interface GigabitEthernet0/1 ip address 172.24.200.5 255.255.255.0 duplex auto speed auto media-type rj45 h323-gateway voip interface h323-gateway voip id GK ipaddr 172.24.200.5 1719 h323-gateway voip h323-id CUBE h323-gateway voip tech-prefix 3# h323-gateway voip bind srcaddr 172.24.200.5 ! ip forward-protocol nd ! ! ip http server no ip http secure-server ! ! ! ! control-plane ! ! ! dial-peer voice 12 voip description Incoming Dialplan translation-profile incoming strip3# session target ras incoming-called number . dtmf-relay h245-alphanumeric! ! ! gateway timer receive-rtp 1200 ! ! ! ! gatekeeper zone local GK 172.24.200.5 zone remote REMOTEGK cisco.com 172.24.200.6 1719 zone prefix GK 3* zone prefix REMOTEGK 5... zone prefix GK 6... gw-type-prefix 1#* default-technology no shutdown ! ! line con 0 line aux 0 line vty 0 4 password login ! scheduler allocate 2 1000 ! end Remote Gatekeeper/CME version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname REMOTE ! boot-start-marker boot system flash flash:c3845-adventerprisek9_ivs-mz.124-25c.bin boot-end-marker ! enable password ! no aaa new-model voice-card 0 no dspfarm ! voice-card 1 no dspfarm ! ip cef ! ! ! ! ip auth-proxy max-nodata-conns 3 ip admission max-nodata-conns 3 ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! archive log config hidekeys ! ! ! ! ! ! ! interface GigabitEthernet0/0 ip address 172.24.200.6 255.255.255.0 duplex auto speed auto media-type rj45 h323-gateway voip interface h323-gateway voip id REMOTEGK ipaddr 172.24.200.6 1719 h323-gateway voip h323-id CME h323-gateway voip tech-prefix 1# h323-gateway voip bind srcaddr 172.24.200.6 ! interface GigabitEthernet0/1 no ip address shutdown duplex auto speed auto media-type rj45 ! interface Service-Engine3/0 no ip address shutdown ! ip forward-protocol nd ip route 0.0.0.0
Re: [OSL | CCIE_Voice] MGCP failover and call preservation
Yes, using debug mgcp packet I see the gateway send a RSIP, the backup CUCM sends it's RQNT, I can then see the AUEP for each channel, and the corresponding AUCX for the active channel. A show ccm-manager shows it as registered to the backup. The call is eventually disconnected when the gateway sends a DLCX to call manager after about 15 seconds. I am testing by shuting down the primary cucm call manager service. On Aug 10, 2010, at 12:57 AM, Daniel Berlinski wrote: Hi Matthew I'm going to test this to check if I can reproduce your fault but a few questions first: Can you confirm to us that your gateway re-registers successfully to the backup CUCM? After registering you should see and AUEP and AUCX coming from CUCM interrogating the calls the gateway preserved. Are you using debug mgcp packets to view these? So, if this is not happening maybe you have got something blocking communication there? How are you testing your failover? By stopping the service in CUCM or by a null route/ACL? Cheers On Mon, Aug 9, 2010 at 3:08 PM, Matthew Hall 1.matt.h...@gmail.com wrote: Not the question you might be thinking. It seems really basic, but I must be forgetting something. When I preserve a call to the PSTN, once mgcp has lost connectivity to the primary call manager, If I hang up the voip phone, the PSTN line does not hang up. I don't see UCM send anything to the MGCP gateway in my debugs and consequently a disconnect never get sent to the PRI. The call eventually disconnects due to timeout. This occurs whether the primary call agent comes back online or not. What am I missing? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] MGCP failover and call preservation
Not the question you might be thinking. It seems really basic, but I must be forgetting something. When I preserve a call to the PSTN, once mgcp has lost connectivity to the primary call manager, If I hang up the voip phone, the PSTN line does not hang up. I don't see UCM send anything to the MGCP gateway in my debugs and consequently a disconnect never get sent to the PRI. The call eventually disconnects due to timeout. This occurs whether the primary call agent comes back online or not. What am I missing? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lan QOS Scenario
FYI, I think this is too complex to answer simply: mls qos queue-set output 1 threshold 2 40 60 100 200 This line gives threshold 1 40% of the buffer space and threshold 2 60% of the buffer space, total buffer space reserved is 100% and max (threshold 3) is 200%. Reserved at 100% tells the queue to not share any of it's buffers with the other queues. Max at 200% says it can borrow up to 2 times it's buffer space from the shared pool (space available in queues 1, 3 and 4). Threshold 1 is set at 40% of the buffer size (that's calculated from the full buffer allocation of the queue), and threshold 2 is set for 60%. Remember though, these don't have to total 100%, in fact you can set threshold 1 to 300% or 1000% if you want. By setting threshold 1 to 40% you tell it to start dropping at that threshold, it has no affect on threshold 2's drop percentage. By setting threshold 1 to 40 and threshold 2 to 60, you are affectively limiting queue 2 to 60% of it's total buffers (unless you assign something to threshold 3). I would say that unless you were asked to mess with the other thresholds, leave threshold 1 at 100. I would also set threshold 3 (max) to 100% of buffers for this question. Because this potentially allows other COS to dou ble the effective size of queue 2, thus turning your 60% threshold into a 30%. But maybe I'm over thinking it. So my answer would look like this: mls qos queue-set output 1 threshold 2 100 60 100 100 sets threshold 1 to 100 percent of buffers sets threshold 2 to 60 percent of buffers (cos 4) sets threshold 3 to 100 percent of buffers Matt On Jun 8, 2010, at 10:37 PM, Pavan wrote: Looks good as farvas i can tell. Normally you would also enabl priority-queue on the interface Sent from my phone On Jun 8, 2010, at 8:20 PM, jammer jones jammerjone...@gmail.com wrote: Trying to understand this a little better. Cisco's documentation is not written in very clear english. Very frustrating trying to understand the threshold values as well as the shape versus share bandwidth values. QOS. Cos 5 for queue 1 queue 2 queue 3 queue 4 0 similar to lab 2 . Queue one has the 25% of the bandwidth. other bandwidth is shared as 30 40 30. If the queue 2 is saturated by 60% then the cos 4 has to be dropped. Here is what I think it is. Can someone please correct me if i am wrong and provide any positive feedback. ! mls qos srr-queue output cos-map queue 2 threshold 2 4 !maps cos 4 to queue 2 and threshold 2 mls qos srr-queue output cos-map queue 4 threshold 1 0 !maps cos 0 to queue 4 mls qos queue-set output 1 threshold 2 40 60 100 200 ! when queue 2 threshold 2 exceeds 60% cos packets with cos 4 will be dropped mls qos ! ! interface GigabitEthernet1/0/1 description Office_912_lab_a switchport access vlan 48 switchport mode access switchport voice vlan 51 srr-queue bandwidth share 1 30 40 30 ! sets queues 2 - 4 to 30 40 30 srr-queue bandwidth shape 4 0 0 0 ! sets queue 1 to 25% of the link mls qos trust cos spanning-tree portfast ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] [SUSPECTED SPAM] RE: Lab and Language settings
It's true this is a dangerous topic, it can all come down to one guys personal judgement at Cisco for all I know and your lab can be revoked and a life ban instated. There is no legal recourse as far as I know, just the end of your CCIE run. If someone point blank asks me for something on the lab, I just avoid answering the question, period. It is fine however to ask tech questions, like why show policy-map interface doesn't work on the 3560. I also answer things that Ben has publicly answered or are knowledge, everyone knows the blueprint on the lab and things like what IOS is used (Ben answered that one). The wording of the NDA is frightening to be honest though. The entire OSL is a violation if you read it strictly. The lines below would forbid the OSL. Disseminating actual exam content via web postings, discussion groups, chat rooms, study guides, etc. Giving or receiving assistance of any kind from anyone for any electronic or lab examination. What if someone sends you a some practice labs or asks you a question that you didn't KNOW was on the lab, you wouldn't know until it was too late. Not to mention that anyone who takes practice labs from vendors will know that there are similarities to the actual stuffheck there have to be, I mean there are only so many ways to setup an mgcp gateway. Where do you draw the line. Best to just be careful, use your best judgement. If it feels like cheating, it maybe. I think cisco is after the cheaters, not people trying to legitimately practice in group settings like we are here. On Jun 5, 2010, at 9:14 AM, wolfsrudel wrote: there's likely enough fundamentalism on the matter to adding other item on the list. let's be less radical and simply ignore such questions and/or comments. for sure, it would best not to forbid than to live and let live. On 6/5/10, Angel Perez gorr...@hotmail.com wrote: Your are right, NDA affects those candidates who have attempted the lab, anyway, please for these people under NDA don't answer any question regarding the lab http://www.cisco.com/web/learning/downloads/guest/learning/c644/ccmigration_09186a00803641d2.pdf http://www.cisco.com/web/learning/le3/ccie/exam/violation_rules.html Thanks Subject: [SUSPECTED SPAM] RE: [OSL | CCIE_Voice] Lab and Language settings From: r.ochi...@mfient.com To: gorr...@hotmail.com; jon1...@hotmail.com CC: siddas...@gmail.com; ccie_voice@onlinestudylist.com Date: Fri, 4 Jun 2010 22:33:43 +0300 It isn’t true that I cannot use the word lab….i can ask what the temperature is like in the lab, is the proctor in the lab, what is the lab topology like without necessarily breaking the NDA. You can ask anything, It’s upon me the person restricted by NDA to tell you that I cannot answer that as I’ll be breaking NDA I think NDA would apply to those who’ve attempted or passed the lab. Others have not agreed to any NDA From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Angel Perez Sent: Friday, June 04, 2010 9:37 PM To: jon1...@hotmail.com; siddas...@gmail.com; osl osl Subject: Re: [OSL | CCIE_Voice] Lab and Language settings Don't worry, just think that if you include the word lab in you question you would be breaking NDA :( From: jon1...@hotmail.com To: gorr...@hotmail.com; siddas...@gmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Lab and Language settings Date: Sat, 5 Jun 2010 03:19:11 +0900 Thanks, and sorry didn’t really mean to ask contents, more of a rough info. question as in blueprints don’t say it, so was pretty much curious. Thanks for the heads up From: Angel Perez Sent: Saturday, June 05, 2010 2:25 AM To: siddas...@gmail.com ; jon1...@hotmail.com ; osl osl Subject: RE: [OSL | CCIE_Voice] Lab and Language settings Hi Jon, you can't ask anything about exam contents, sorry From: siddas...@gmail.com To: jon1...@hotmail.com; ccie_voice@onlinestudylist.com Date: Fri, 4 Jun 2010 15:39:27 +0100 Subject: Re: [OSL | CCIE_Voice] Lab and Language settings No, I don’t think so.. As a rule of thumb just select US (English) where ever needed. Ash From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jon1992 Sent: 04 June 2010 15:36 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Lab and Language settings Hi Was just curious, does the lab get involved at all with locale settings or how to upload files for the languages to be available? I didn’t see much mention of it in the blueprint, so was curious for those who have attempted, any mention of it? Thanks Jon Hotmail: Powerful Free email with security by Microsoft. Get it now. Hotmail: Trusted email with powerful SPAM protection. Sign up now.
Re: [OSL | CCIE_Voice] Shared lines in CME SRST
No privacy on, but I'll give the gateway are reboot next time and see what happens. matt On Jun 6, 2010, at 7:06 AM, Angel Perez wrote: Hi: Do you have privacy on at any of the phones before going to srst? Also sometimes you have to reload the gw with cme srst to make it works properly hth From: 1.matt.h...@gmail.com Date: Sat, 5 Jun 2010 22:55:23 -0500 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Shared lines in CME SRST Problem I'm having is as follows: CME SRST Two phones have a shared line 2010 I have srst dn mode set to octo, when registering, one or both of the phones always come up as remote in use and stay that way, no matter how many times I get them to unregister and reregister. Anyone else seen this before? Thanks Matt ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Shared lines in CME SRST
Problem I'm having is as follows: CME SRST Two phones have a shared line 2010 I have srst dn mode set to octo, when registering, one or both of the phones always come up as remote in use and stay that way, no matter how many times I get them to unregister and reregister. Anyone else seen this before? Thanks Matt ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Show Gatekeeper Calls - Odd Issue
In my experience changing the call manager service parameters for default inter and intra region codecs to g729 causes this to work in both directions. As long as your dial-peer is g729 and the GK trunk is in a g729 region. Matt On May 21, 2010, at 12:06 PM, Graham Hopkins wrote: Matthew - I found that the call from HQ to BR is fine and shows 16kbps, its the call from BR2 to HQ that has the problem, so to complete the task just make the call in the right direction :-) Did you have the same issue ? Regards Graham Hopkins On 21 May 2010, at 16:50, Berry, Matthew J. wrote: That’s great to know. I burned a few hours last night on Proctor trying to get this to work. Hopefully we won’t be asked a question like that on the lab. According to my understanding, then, we cannot technically complete and get points for question 5.1 since it requires you to produce the “show gatekeeper calls” output listed in the question. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.com From: Roger Källberg [mailto:roger.kallb...@cygate.se] Sent: Friday, May 21, 2010 10:44 AM To: Berry, Matthew J.; CCIE Voice OSL (ccie_voice@onlinestudylist.com) Subject: RE: [OSL | CCIE_Voice] Show Gatekeeper Calls - Odd Issue Also this, http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg16188.html Roger Källberg Unified Communication Consultant Cygate AB From: Berry, Matthew J. [mailto:mjbe...@krollontrack.com] Sent: den 21 maj 2010 17:21 To: CCIE Voice OSL (ccie_voice@onlinestudylist.com) Subject: [OSL | CCIE_Voice] Show Gatekeeper Calls - Odd Issue All – I had an issue last night on Vol 2 Lab 2. I am sending calls from HQ (Region = HQ) to BR2 over my H.225 trunk (Region = GK). Region setting between HQ and GK specifies G.729. I have a transcoder registered on the BR2 router. When I call across the gatekeeper, my endpoints show G.729, but “show gatekeeper calls” shows 128kbps. Extremely odd. Does anyone have insight into this? Thanks! Matthew Berry, CCVP, Sr. Unified Communications Engineer Kroll | 9023 Columbine Road, Eden Prairie, MN 55347 Single Number Reach +1 952 516 3748 | Fax +1 952 516 3646 | mjbe...@kroll.com www.krollontrack.com | www.kroll.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] translation rule
You are correct, this only catches when there is a null number, so : voice translation-rule 100 rule 1 /^$/ /42000/ If applied to an ANI would only apply to calls that had no ANI set and rewrite it to 42000. Matt On May 21, 2010, at 3:58 PM, Ashar Siddiqui wrote: Thanks David I have already gone through that document many times :) Thanks Wael for your explanation. I was actually thinking that what would a null number be in my example. I have this customer router which has this specific rule for inbound calls. Will it work when PSTN will send null digits? doesn't make sense to me. What is a null/unknown digit? Ash David Holman wrote: I keep this link handy for voice translation questions: http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080325e8e.shtml On Fri, May 21, 2010 at 4:15 PM, Wael Agina waelag...@gmail.com wrote: Dear Ashar, The ^$ is catching null, which could be used to catch calls from unkown. example usage, drop any calls from PSTN that has ANI of unkown type. On H323 you could use following rule to do this voice translation-rule 1 rule 1 reject /^$/ voice translation-profile Drop-Unknown translate calling 1 dial-peer voice 1 pots direct-inward-dial incom called . call-block translation-profile incoming Drop-Unknown For you example may be it i setting unknown ANI to be 42000 for example, bu not sure, need to be tested. Regards, Wael Agina On Fri, May 21, 2010 at 11:02 PM, Ashar Siddiqui siddas...@gmail.com wrote: Hi, I know I may sound stupid to some but I really want to know the purpose of ^$ in a translation rule for e.g: voice translation-rule 100 rule 1 /^$/ /42000/ ! ^$ is null...what does it mean? what is a null number? Ash ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Thanks and Best Regards, Wael Agina ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Dial Plan Testing Strategy ? (Mike Brooks)
Mike, I used to just test at the end, but I found it to be counter productive if I made several mistakes, I now test as I go, with the exception of aar, it only takes a few minutes more to test each one as I go, shutting down a voiceport takes only seconds, as does bringing back up. I makes me more confident in my foundation I build the dial plan as well. Test it all at the end was easier when I used to chart out my full dial plan on notepad (as I did in V2), but now I just build as I go and it's much harder to test the whole dialplan at the end without a table. Matt What is everyones thoughts on testing dialplan ? If you have multiple paths for calls such backup gateways, or teho with backups etc, would it make sense to get through dialplan and then just test at the end. Or test as you go, both the primary and backup path after implementing each route pattern ? Testing as you go would involve shutting down the voice-port on the gateways after each pattern is implemented and verifying it works before moving on to the next bullet/task. Just curious what you all feel is the best approach. Thx, Mike ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Invoking CUC Call Handler Transfer Rules
I used to know ho to invoke these, rather than having the call go to greeting. They are normally invoked when you transfer from the AA I thought. I created a call handler with an extension 2401. I set the alternate transfer rule to enabled, set it to no greeting and to forward to an extension (2001) with a direct release to switch. When I enter the AA I dial 2401, and it tells me to wait while it transfers, and proceeds to give me the greeting and go into the mail box of 2401. My goal is to do this: Dial 2401, have 2401 transfer directly to 2001. So essentially I need to first find out why the AA isn't getting to the transfer rule, then find out how to invoke the transfer rule without the aa. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
