Archive for the ‘exam’ tag
JNCIS – Epic Win!
I quit my job…by design. I start a new gig on Tuesday and am getting back to the world of Cisco. As a last nod to Juniper, I decided to use an exam voucher I had and take the JNCIS-ENT exam. Easy pass.
The content was right along with the exam objectives, so there were no surprises. Most of the topics are things I’ve done a thousand times on the job. There were some things, though, that were beyond my experience. IS-IS was the big one. The very first question I got was about IS-IS metrics, and I had absolutely no clue what the answer was. Nor did I have any clue about the other IS-IS questions. I went 0-for-3 on those guys. The only other problematic topic was HA, which didn’t really surprised me. I was able to answer the VRRP questions, but I’ve never done any GRES, ISSUe, RTG, etc., at any point in my career. It wasn’t surprising that I didn’t do too well on those. Everything else was cake, and I only missed 6 questions in my comfort zone.
The exam was yet another top-notch effort from Liz and the group, but there was one questions that didn’t meet the standard set by the others. It was a VRRP question, but it used some awkward wording that that I read over and over. I just used the context of the questions to give an answer and moved on.
There was really nothing else to report. It was a great exam, so don’t be afraid to take it if it’s next on your list.
Send any Cisco refresher courses questions my way.
JNCIA – Epic Win!
Maybe not epic, but a win nonetheless.
My boss is over all the network guys in the company, and that includes guys that support different divisions and departments. He told me he was tired of waking up at 2am every morning to fix a problem the other groups can’t handle, so he’s working to get the junior guys motivated to learn for themselves. One technique he’s implemented is to force them to get their CCNAs and JNCIAs by June. Since he made it part of the job description, that means that everyone above the Analysts has to meet those requirements, too. I made the deadline with plenty of time to spare.
Do you remember the full day off of work I had to take to sit the CCNP exams? The 2-hour drive to a prison town, lunch, a 2-hour exam, and 2 hours back? That sucked. I live in a major metropolitan area now, so my travel time to the nearest testing center is 45 seconds. I mean, literally 45 seconds. It’s right across the street from my apartment complex. Easy walk if it wasn’t so cold. That’s good, too, because I showed up this morning, and the center didn’t have any power! Someone plugged in a coffee maker in the break room, and power went out in a whole wing of the building. Since I always get there early, I was actually able to drive home, wait for them to fix the problem, and still be there at my scheduled time. Convenient for sure.
I must say that the exam was pretty darn good. It may, in fact, be the best IT exam I’ve ever taken. The breadth of material was awesome; it had questions from the absolute basics to some of the stuff I saw on the CCIE R&S written. Since I’ve been doing networking for so many years and have my string of certs, the exam was pretty easy to me, but I’m sure an absolute network newb would find the material’s scope a little overwhelming. The exam scores very high on the fairness meter, as well. The questions were clearly written; the exhibits were legible and well-marked. Best of all, there were no real trick questions. They asked what they wanted you to answer and provided the answer to you. There were no assumptions or judgments involved in trying to figure out what was being asked.
Overall, I was very impressed with the exam. Two thumbs up. I can only hope the rest of the exam in the track are this good. I won’t know until after I pass my CCIE lab, though. :)
Send any certification delays questions my way.
CCIE R&S Written – Epic Fail
It’s been a long time, eh? I’ve spent the last month or so with my nose down in a book and my mouse in a Google+ Hangout window studying my rear off for the CCIE R&S Written. Too bad I didn’t pass it.
The exam consisted of 77 questions over a 2 hour window. That’s plenty of time to finish; I think I had 48 minutes left when I was through, so time wasn’t a problem. There were only 2 or 3 questions where I was totally lost, so the technology wasn’t a problem. The big problem, like always, was the usual crap questions that are in these exams. Some didn’t provide all the required information. Some were impractical examples of deployments you would never use in the field. Some were on deprecated technologies. Hell, I had one that involved CatOS. Really? CatOS? Since I only failed by about 2 questions (like I always do), these shenanigans are magnified in my mind. It really irks me how these exams are being done; foggy questions don’t really measure ability.
I did have one great advantage last week that I’ve never had – I took the exam at Cisco Live and had 489247248 CCIEs around me willing to help. Since I took the exam on Sunday, I was able to ask people face-to-face for advise on what I need to do to pass, and the consensus was that I needed to practice how to answer questions the way Cisco wants you to answer them since the material wasn’t really that hard. The suggested next steps ran the gamut, too. Some suggested just firing from the hip for answers – the whole “your first guess is always right” theory. Others suggested that I just brute force the exam. Still others even suggested brain dumps along with the idea that we’ve all put in our time and effort already and that the terrible questions shouldn’t be what’s holding us back.
You guys know me by now. I’m definitely not going to give up or anything stupid like that. I’ll probably take a week off to recover from Cisco Live and then head back to the studies. I’ll do the usual brute force method, but I’m going to grab a copy of the Boson exams (which were also suggested) to supplement. I would guess that I’ll try again around the first of August, but we’ll see.
Send any beatin’ sticks questions my way.
IIUC Update – Passed!
I passed the IIUC yesterday, so now I'm a CCNA Voice. It's kind of belittling to get a CCNA-level certification at this point in my career, but I didn't want to be completely left behind, so I figured I should move into some voice stuff before I'm left in the dust.
The exam was probably the best Cisco exam I've ever taken. Of all the exams I've taken in the last few years, this is the only one that didn't have questions with huge misspellings or grammatical errors. I was really taken aback at that since a good portion of the questions from some of my recent CCNP exams were plain unreadable. I think I remember leaving a comment on one IIUC question about the word "an" being left out of a description, but that was no big deal. I'm not that obesessive-compulsive.
The exam wasn't that difficult, either. There were some questions that I wound up reading over and over again because the answer was just so obvious to me, and I wanted to make sure I didn't miss something. There were a good number of questions on the CCA which I didn't enjoy one bit. Those are the questions that ask "what tab do you select to do X?" Those, in my opinion, are pointless memorizations since, outside of the exam, I can just click around to figure out what's being asked. On that same vein, I did get a simulation to configure a UC500 to meet requirements, but that was pretty trivial. The most difficult questions dealt with connecting your CME to either the PSTN or PBX; I actually focused my studies on that specifically, but I still blew it. Oh, well.
Now that voice is finished, I'm back on the R&S track and am studying for the CCIE written. I'll get around to marking the calendar for my attempt, but, for now, I'm going to take a few days off and enjoy some cigars.
Send any Gilligan's Island episodes questions my way.
Audio commentary
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IIUC Notes – Inbound Dial Peer Matching
More IIUC notes. As always, feel free to correct as needed.
To match inbound calls to a dial peer, CME (and CUCM?) uses the following steps.
- Match DNIS (the dialed number) with the incoming called-address config in the dial peer
- Match the ANI (the calling number or caller ID) with the answer-address config in the dial peer
- Match the ANI with the destination-pattern config in the dial peer
- Match an incoming POTS call to the port config in the dial peer
- Match dial peer 0
Matching dial peer 0 is bad, and it took me an inquiry on Twitter and a buddy to realize why. Here are a few highlights as to why. I believe the full scope of the badness of dial peer 0 is really beyond the IIUC exam.
- It takes whatever codec is sent to it and can't be hard-coded.
- DTMF is sent in the audio stream, so, if you wind up with a G.729 or other highly-compressed codec, you may have problems getting DTMF across successfully.
- IP precedence values are stripped out of the packets, so it's just plain data now.
- RSVP is disabled.
- No application support, so you can't do IVR. [Will AA work?]
- No DID support. This means the wife can't dial your desk with your published number.
Stubby Post – Changes to CCNA Voice, CCVP, and CCSP
I don't usually cover news from Cisco, but they've changed some certification stuff around again, and I thought I would bring it up. This time they've changed the CCNA Voice, CCVP, and CCSP, so, if you've on those tracks, be careful what you're studying!
CCNA Voice
Circle 28 February 2011 on your calendars. That's when the CCNA Voice track gets a shakeup. The IIUC (640-460) exam will be no more, and passing CVOICE (642-436) will no longer be a valid way to get the cert. After the big day, you'll have to take ICOMM (640-461). This seems to be a much broader exam instead of having the enterprise and commercial focuses in CVOICE and IIUC, respectively. Look out for both CME- and CUCM-based topics including a troubleshooting section.
See also:
Wendell Odom's blog at NetworkWorld
CCVP
The CCVP is now known as the CCNP Voice. There are still five exams to get the certification, so it's not that different. The QoS exam is gone, but the new CVOICE (642-437) exam includes QoS, so keep studying those queueing methods. The TUC exam is replaced by TVOICE (642-427), which, on the surface seems to be just an update. The CIPT1 (642-447), and CIPT2 (642-457) exams also look like they're simply updated, but you'll have to ask a Voice guy since I don't really know the differences here.. The last exam is CAPPS (642-467), and covers Unity, VPIM, and Presence. Fun stuff.
See also:
Wendell's blog again
CCSP
Like the Voice track, the CCSP gets a name change and is now known as the CCNP Security. There are still four tests like the old track, but the content is updated. You have to take the SECURE (642-637), FIREWALL (642-617), VPN (642-647), and IPS (642-627). Word on the street is that the new VPN exam eliminates the inconsistencies with VPN deployment methods taught in SNAF and SNAA.
See also:
Wendell's blog again
Can someone explain why CCSP and CCNP Security are both still listed on the professional cert page at Cisco, but the CCNP Voice gets a "formerly known as" moniker?
CME Exercise #1
I tried something like this earlier this year with STP. It got rave reviews (from my mother), so I figured I try it again.
Below is a list of requirements for configuring a router as a call processor. In a lab or in your head, configure the router to support the features as listed. This isn't a contest or anything like that. If you get it right, a virtual thumbs up is all I can afford to give you. There are some licensing issues for running this stuff in GNS3/dynamips, so I can't help you out on that. I'll just hint that GNS3 and dynamips will bind to real networks and that copies of a compatible IP softphone are available.
Here we go.
- Telephony
- Maximum of 10 DNs
- Maximum of 5 ephones
- DHCP server that provides the appropriate DHCP scope option for getting the phones online
- Phones
- Phone 1
- Sales Phone A
- Button 1: extension 1001
- Button 2: intercom to phone 3 labeled as "Lackey"
- Pickup Group 3001
- Phone 2
- HR Phone A
- Button 1: extension 1002
- Pickup Group 3001
- Phone 3
- Sales Phone B
- Button 1: extension 1003
- Button 2: monitor button 1 on phone 1
- Button 3: intercom to phone 1 labeled as "Boss" that answers unmuted
- Pickup Group 3002
- Phone 1
- Paging
- Each department should have its own paging group.
- All phones should be in a paging group for broadcasting emergencies to all employees.
- Call Parking
- 2 call parking DNs
- 1 CP DN should be dedicated to phone 2
- Music on Hold
- MOH should play when a user is on hold or in a park slot
- After-hours
- After hours should be Mon – Fri from 7pm to 7am
- No one should be able to dial 1003 after hours
- No one should be able to dial 1002 any day at any time
I'll get my own answer together and post the consensus result in a few days. In the meantime, let me know how terribly I did.
Send any unlicensed CIPC phones questions my way.
IIUC Notes – Voice Ports and Dial Peers
More of my IIUC study notes. As always, feel free to correct. I really need to have a real post, don't I?
show voice port summary
- Shows the voice ports available for use
R1#show voice port summary
IN OUT
PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC
============== == ============ ===== ==== ======== ======== ==
50/0/1 1 efxs up up on-hook idle y
50/0/1 2 efxs up up on-hook idle y
50/0/2 1 efxs up up on-hook idle y
50/0/2 2 efxs up up on-hook idle y
50/0/3 1 efxs up up on-hook idle y
50/0/4 1 efxs up up on-hook idle y
50/0/5 1 efxs up up on-hook idle y
- An ephone-dn shows up as efxs, so all these are ephone-dns.
- Channels are numbered 0-23; timeslots are numbered 1-24
FXS Ports
- Connect to end stations like analog phones and fax machines
- Signaling
- Ground start: New connections started by grounding wires
- Typically used when tied to PBXes
- Loop start: New connections started by sending DC voltage
- Default
- Typically used when connecting to analog devices
- Ground start: New connections started by grounding wires
- Call progress tones
- Audible tones to let the user know the status of a call
- Dial tone, busy, call waiting, etc.
- Different in each geographical area
- Audible tones to let the user know the status of a call
- Caller ID
- Identifies the name and number that calls on this line should appear
R1(config)#voice-port 0/0/0
R1(config-voiceport)#signal loopStart <- Use loopstart signaling
R1(config-voiceport)#cptone PE <- Uses CP tones from Peru
R1(config-voiceport)#station-id name Corporate Fax
R1(config-voiceport)#station-id number 5551212
FXO Ports
- Connects to CO or PBX
- A lot of the same configurations as FXS ports
- Two additional to discuss
- dialt-type: DTMF or pulse dialing
- ring: The number of rings to wait before answering; usually 1
- Think of allowing a home user to answer the phone before the fax machine picks up
R1(config)#voice-port 0/0/1
R1(config-voiceport)#dial-type dtmf <- touch tone
R1(config-voiceport)#ring 3 <- wait 3 rings before answering
Digital Voice Ports
- Unlike analog voice ports, digital voice ports must be configured to function with the network to which they are attached.
- Voice and WAN interface cards (VWICs) provide digital voice port
- show controllers t1
- Framing: defines how to format the frames
- SF or ESF
- Line coding: encodes the signal in a way to maintain sychronization
- AMI or B8ZS
- Clock source: defines who dictates the clocking
- Signaling: channel signaling
- CAS: use ds0-group
- Ports show up as 0/0:1, where 0/0 is the physical port and 1 is the ds0 group
- CCS: use pri-group
- Ports shows up as 0/0:23, where 0/0 is the physical port and 23 is the signaling channel (16 in E1)
- CAS: use ds0-group
R1(config)#isdn switch-type primary-5ess <- If using CCS
R1(config)#controller t1 0/0
R1(config-controller)#framing esf
R1(config-controller)#linecode b8zs
R1(config-controller)#clock source line <- get clocking from provider
For CAS:
R1(config-controller)#ds0-group 1 timeslots 1-24 type fxo-loop-start <- Using FXO loopstart signaling
-or-
For CCS:
R1(config-controller)#pri-group 1 timeslots 1-24 <- assumes signaling from CCS and ISDN switch-type
Dial Peers
- "Routing" for phone numbers
- Tells a voice gateway where to send calls based on dialed number
- Two types dial peers
- POTS: Traditional connections like T1 and analog phone lines
- VOIP: Connections to an IP address
- show dial-peer voice summary
R1(config)#dial-peer voice 1101 pots
R1(config-dial-peer)#destination-pattern 1101 <- This number…
R1(config-dial-peer)#port 0/0/0 <- …is on this FXS port.R1(config)#dial-peer voice 1102 pots
R1(config-dial-peer)#destination-pattern 1102 <- This number…
R1(config-dial-peer)#port 1/0:23 <- …is on this T1 PRI port.R1(config)#dial-peer voice 1103 voip
R1(config-dial-peer)#destination-pattern 1103 <- This number…
R1(config-dial-peer)#session target ipv4:10.10.10.1 <- …is at this IP address…
R1(config-dial-peer)#codec g711ulaw <- …and use this codec when you get there.
IIUC Notes – More Phone Features
Here are some more notes from my IIUC studies. As always, corrections requested.
- Broadcasts messages to a group for a one-way communication
- Paging groups are used to limit which phones get the broadcast
- Paging can be unicast or multicast
- Unicast groups limited to 10 members
- Multicast requires mcast support on the network
- Paging configurations can be unicast, multicast, or multiple-group
! Unicast Paging
! When 1044 is dialed, ephone 1 is paged
R1(config)#ephone-dn 44
R1(config-ephone-dn)#number 1044
R1(config-ephone-dn)#paging
R1(config-ephone-dn)#exit
R1(config)#ephone 1
R1(config-ephone)#paging-dn 44! Multicast Paging
! When 1045 is dialed, ephone 2 is paged
R1(config)#ephone-dn 45
R1(config-ephone-dn)#number 1045
R1(config-ephone-dn)#paging ip 239.1.1.100 port 2000
R1(config-ephone-dn)#exit
R1(config)#ephone 2
R2(config)#paging-dn 45! Multiple Group Paging
! When 1046 is dialed, both ephones 1 and 2 are dialed
R1(config)#ephone-dn 46
R1(config-ephone-dn)#number 1046
R1(config-ephone-dn)#paging group 44, 45
- There is a limit of 10 DNs in the paging group.
After-hours Call Blocking
- Allows you to configure time ranges and patterns that cannot be called during those ranges
- Three steps
- Defines days and/or hours that are considered after-hours
- Specify patterns to be blocked
- Create exemptions
R1(config)#telephony-service
R1(config-telephony)#after-hours day mon 18:00 07:00 <- afterhours = 6pm to 7am
R1(config-telephony)#after-hours day tue 18:00 07:00
R1(config-telephony)#after-hours day wed 18:00 07:00
R1(config-telephony)#after-hours day thu 18:00 07:00
R1(config-telephony)#after-hours day fri 18:00 07:00
…
R1(config-telephony)#after-hours date Dec 25 00:00 00:00 <- Christmas is after hours
…
R1(config-telephony)#after-hours block pattern 1 91900……. 7-24 <- Pattern index 1 blocks 900 numbers 7day/24hours
R1(config-telephony)#after-hours block pattern 2 91………. <- Pattern index 2 block all long distance after hours
…
R1(config-telephony)#login timeout 15 clear 18:00 <- Allows logins for entering a PIN for after-hours exemption; times out in 15 minutes and clears at 18:00
R1(config-telephony)#exit
R1(config)#ephone 1
R1(config-ephone)#after-hours exempt <- the boss's phone can call anywhere except the 7-24 patterns
R1(confg-ephone)#ephone 2
R1(config-ephone)#ping 1234 <- Your phone can log in with this PIN for after-hours access
- Phones have to be restarted or reset for the Login key to be enabled.
Call Accounting
- It's important to see who is calling international numbers every day at lunch.
- Call Detail Records (CDRs) record who called what number when for how long plus more stuff.
- CME logs CDRs to the logging buffer, syslog, or both.
- Logging buffers clear when a router loses power, but it may be better than nothing. <- Don't do this ever! Get a syslog server!
R1(config)#logging buffer 512000 <- Set the logging buffer size to 512000 bytes
R1(config)#dial-control-mib retain-timer 120 <- Roll records out in 120 minutes
R1(config)#dial-control-mib max-size 100 <- Only keep last 100 records
- Sending to syslog allows you to keep more records
R1(config)#gw-accounting syslog
R1(config)#logging 192.168.0.2 <- Log to this server
- Account codes are used for billing.
- Each department or unit can enter a code that appears in the CDR for use later.
- Users press the Acct key when the call is ringing or connected to enter their code.
Music on Hold
- Do I have to explain what MoH is?
- WAV or AU file in flash
- Files must be G.711 or G.729
- G.711 is recommended since it is of higher quality
- Can be delivered via unicast or multicast
R1(config-telephony)#moh piratedmusic.au <- Plays a local audio file as MoH
R1(config-telephony)#multicast moh 239.1.1.15 port 2001 <- multicast the MoH
IIUC Notes – Phone Features
Here are some more notes from my IIUC studies. As always, corrections requested.
Local Directory
- Allows users to look up names
- Allows names to show up when dialing or receiving a call
- Most phones have a directory button; some have a menu options for the directory
R1(config)#ephone-dn 1
R1(config-ephone-dn)#name Roger Smith
- Directory entries can be added manually
R1(config-telephony)#directory entry 1 1700 Corporate Fax
R1(config-telephony)#directory entry 2 1701 HR Fax
- By default, sorting is done alphabetically by first name.
- Sorting can be changed
R1(config-telephony)#directory last-name-first
Call Forwarding
- Can be done by the user or through CLI
- User presses CFwdAll button, enters a number, and #; pressing CFwdAll again cancels forwarding.
- CLI forwarding is more flexible
R1(config-ephone-dn)#call-forward busy 1800
R1(config-ephone-dn)#call-forward noan 1800 timeout 25 <- if no answer after 25 seconds
R1(config-ephone-dn)#call-forward max-length 0 <- disabled forwarding
R1(config-ephone-dn)#call-forward max-length 4 <- restricts forwarded number to a length of 4 digits
- H.450.3: A voice gateway redirects the forward to another gateway instead of using the phone as a proxy
- Direct path from originator to destination
- Frees up network resources by keeping path direct
- Keeps latency and jitter down by avoiding long looping paths and a hairpin turn at the phone
- Forwarding patterns can help restrict where calls can be forwarded
R1(config-telephony)#call-forward pattern 1… <- allows forwarding to a 4-digit number starting with 1
Call Transfer
- H.450.2: A voice gateway redirects transfers to another gateway instead of using the phone as a proxy.
- The user doing the transfer is dropped from the conversation after transfer is complete.
- Generically, there are two types of forwarding.
- Blind: sends the caller to the number blindly
- Consult: allows you to talk to the endpoint before transferring the call
- CME has three types of forwarding.
- full-blind: blind transfers using H.450.2 or SIP REFER
- full-consult: consult transfers using H.450.2 or SIP REFER if second line is available; if not, fall back to full-blind
- local-consult: Cisco-proprietary method for full-consult
R1(config-telephony)#transfer-system full-consult
- or -
R1(config-ephone-dn)#transfer-mode consult
- Transfer patterns work similarly to forwarding patterns
R1(config-telephony)#transfer-patter 1…
Call Park
- Call parking allows a user to retrieve a call from any phone by "parking" the call to an extension.
- The call can be picked up from any phone able to dial that extension.
- Park numbers can be assigned randomly or manually.
R1(config-ephone-dn)#park-slot <- makes this DN a park slot
- Call parking has several options.
- reserved-for dn: Only that DN can use this park-slot
- timeout seconds: Ring the phone phone that parked the call after that many seconds to remind them of the park
- limit count: After that many timeout intervals, drop the call. Not good for customers.
- notify dn [ only ]: Notify that DN when a timeout is reached
- recall: Sends the call back to the original phone when the timeout is reached
- transfer dn: Sends the call to this DN when the timeout is reached
- alternate dn: If the DN in the transfer command is not available, go here
- retry seconds: Try to transfer again after this many seconds
- The phone must be reset for call parking to take effect.
Call Pickup
- Allows users to pick up other ringing phones
- Best to use pickup groups so the sales guys don't pick up support calls by accident
R1(config-ephone-dn)#pickup-group 5000
- There are three methods to pickup a call.
- Directed pickup: A user picks up a ringing phone by pressing PickUp followed by the target DN.
- Local group pickup: A user picks up a ringing phone in his pickup group by pressing GPickUp then *.
- Other group pickup: A user picks up a ringing phone in another pickup group by pressing GPickUp then the other group number.
Intercom
http://www.youtube.com/watch?v=3P2dbwrT_fQ
- Technically is a speed dial and auto-answer combination
- Intercom button is pressed, which dials a DN bound to another phone; that phone automatically answers on mute.
- The DNs involved usually (?) can't be dialed.
- e.g., A101
R1(config)#ephone-dn 99
R1(config-ephone-dn)#number A99
R1(config-ephone-dn)#intercom A98 label "Boss"
R1(config-ephone-dn)#exit
R1(config)#ephone-dn 98
R1(config-ephone-dn)#number A98
R1(config-ephone-dn)#intercom A99 label "Lackey"
R1(config-ephone-dn)#exit
R1(config)#ephone 54
R1(config-ephone)#button 5:99
R1(config-ephone)#restart
R1(config)#ephone 73
R1(config-ephone)#button 5:98
R1(config-ephone)#restart
- Other options
- barge-in: Places existing calls on hold on the other end and barges n
- no-auto-answer: Rings instead of auto answers
- no-mute: Doesn't mute when auto answering. Can you say spying?