Share your CVoice v8.0 Experience

Thank you Voicetut!

Can someone send me the latest dumps to take the cvoice v8 exam…. plssss to email planchaman@hotmail.com…..

@ 9tut
Thank for creating a new path for the new exams 😉

Can anyone tell us if there is an LabSim in this 642-437 Exam or not.

Greetings from Germany and GOOD LUCK to all !!

No 1 has given this exam? :O .. waiting for dumps =D .. i heard this exam has simulation as well pls confirm thanks 🙂

can anybody tell me whether ther r simulations in this cvoice exam coz p4s gives labs in the bundle for cvoice exam.

There are no lab style simulations in this exam. Just D&D and multichoice questions.

Are there any video tutorials on this new exam?

Not sure, cbtnuggets is working on it though :

http://www.cbtnuggets.com/series/1089

Have any one share 642-437 exam certification guide link?

I will be taking this exam on 30 July and will be sure to provide some more appropriate feedback to the forum.

i take 642-436 is same as 642-437 v8 Cvoice can complete ccnp voice v8 or must again take 642-437 v8 ?

berma, your comment makes no sense, please rephrase

@berma: 642-436 has been retired. You have to take the 642-437 v8.

I have noticed that it is similar to the 642-436 QOS. So if you know that material very well then you should not have any difficulty with new version.

Hi All,

Does anyone know where i can get the cbt nuggets for voice tracks.

Many thanks in advance

http://hotfile.com/dl/124745772/09c3348/Cisco.Pass4Sure.642-437.v2011-07-19.by.nagamin.145q.vce.html

Please is there someone awake?

1 C
2 Why C is not Correct?
7 Only A or B Possible… is B wrong?
38 What does mean FXS/DID (why not A?)
43 which the difference between B and D
47 which the difference between B and D (i don’t like B, because how can we analyse the missing packets???… interpolation is next and previous )
56 what about Untrusted?

@ValerioVB

1 is A

The Class-Based Packet Marking feature provides users with a user-friendly command-line interface (CLI) for efficient packet marking by which users can differentiate packets based on the designated markings.
CoS is classification of specific traffic by manipulating the class of service bits in the Ethernet frame header whereas IP Precedence and DSCP is configured by changing the TOS Field in IP Header frame.

2 is B

By default, De-Jitter buffer runs in an adaptive mode where it dynamically adjusts to the amount of jitter present up to a point.
The DSP algorithms in the codec take samples throughout the voice call and adjust the value of the average delay as network jitter conditions change.
The size of the jitter buffer is adjusted upward or downward as needed to ensure smooth transmission of voice frames to the codec.
If the jitter is so large that it causes packets to be received out of the range of this buffer, the out-of-range packets are discarded and dropouts are heard in the audio.
For losses as small as one packet, the DSP interpolates what it thinks the audio should be and no problem is audible.
When jitter exceeds what the DSP can do to make up for the missing packets, audio problems are heard.

http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a00800945df.shtml

FXS/DID = 4-Port FXS/DID Voice Interface Card, DID = Direct inward dialing

An FXO trunk is one of the simplest analog trunks available. Because Dialed Number Information Service (DNIS) information can only be sent out to the PSTN, no direct inward dialing (DID) is possible. ANI is supported for inbound calls. Two signaling types exist, loopstart and groundstart, with groundstart being the preferred method. An FXS DID trunk can receive only inbound calls, thus a combination of FXS DID, and FXO ports is required for inbound and outbound calls.

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/srnd/design/guide/gatewy.html#wp1052

7 show call active voice

To display call information for voice calls in progress, use the show call active voice command in user EXEC or privileged EXEC mode.

@nagamin

Dude, thank you for share your CVoice v8.0 Experience!

Please tell me/us: Are all question in the real exam come from your 149q.vce, or
are there a few other new Questions?

Is it possible to pass the Exam with this dump?

I haven’t take 642-437 yet. That is what I got and share you guys.
But I’m sure about 65% of real exam was from question 1 to 66 from this dump.
I’m still working on new version of dump with explanation. I’m not sure how long gonna take.

@ Nagamin
I’m sure you are a good guy… but please don’t think that others guys are not good, as well 🙂
I have same dump from where you simply copied your explanation… but i have some doubts about!

1. TOS = IP packet Field = Layer 3

2. “If the jitter is so large that it causes packets to be received out of the range of this buffer, the out-of-range packets are discarded and dropouts are heard in the audio.” = What does it mean “IF”? OTHERWISE IT’S ADJUSTED DINAMICALLY…. how mach is “so large”?

7. Have you ever tried the command debug voip ipipgw?

38. FXS/DID = 4-Port FXS/DID Voice Interface Card, DID = Direct inward dialing….. SO IT MEANS THAT FXS/DID is not a kind of port…. it’s a CARD, made by port! Which kind of port? FXS!

Can you share you dump for us?
Thanks in advance.

You are welcome guys 🙂

http://www.4shared.com/document/RMLNZdtY/642-437_52.html

Great Valerio !

I find a diffent in the D&D Q39,
your dump says media flow trough, the dump from nagamin says media flow arround.

So what is the correct answer, my favor is media flow through, but i am not really shure.

Greets to you Voiceguys

Yes Media Flow through is correct answer, my dump media flow arround is wrong.

hi guys , i just passed with exact 790 ! Many Thanks to nagamim and Valerio.
i studied using p4s from nagamim and answered the questions the same that have on the dump (i think just 2 i didn´t put the same answer) ,3 questions from AT 5.2 that Valerio provided (72,73 and 74) that don´t have on nagamim dump and almost 5 new questions.

one related to interface/crads that have wink start . pulse/dial tone options and i think the rigth answer is E&M.
i don´t remember the other ones at this time , maybe after i will remember and post here .
Thanks for all.

@Andren
Thanks for update. Way to go man.
If you don’t mind, please let us know your % per section.
It was show you on very end of your result sheet.
So we can trace on dump.

@nagamim, follow :
describe a dial plan 100%
describe the basic operation and components involved in a voip call 50%
implement CUCME to support endpoints using CLI 88%
Describe components of a gateway 60%
implement a gateway 71%
implement cisco unified Border Element 40%
Describe the need to implement QOS for voice and video 50%
Describe and configure the Diffserv QOS Model 82%

hello,

as I can download the book CVOICE v8.0 642-437.

I’m reading the CVOICE 642-436.

these terms and CBWFO LLO
please.

the questions that i remember that had on my exam : following question numbers from AT 5.2:
3 until 12;
14 drag&drop but have 5 boxes to fill instead of 4 so you need to add “default dial-peer”
15
16 i answered wrong, different from the dump
17 until 55
57,58, 59
66
72,73,74
5 new questions…

note that i put the answers from the pass4sure that nagamim provided. i dind´t compared the answers from P4S with AT .

Andren Thank you very Much for your very precious and precise informations… and Congratulations!

Wow.. Thanks alot Andren.
That will help us alot. So you done with CCNP Voice?
Congrats man.

probably yes, just waiting for the cisco update their database regarding that i passed 642-642 qos when i completed the ccip 2 years ago and maybe can replace the CAPPS (642-467). Tomorrow i´ll check on my cisco login certification to make sure.

guys , this can help you on your study 642-437 cvoice quick reference online (some limitations)

http://techbus.safaribooksonline.com/book/certification/ccnp/9780132375580/firstchapter#X2ludGVybmFsX0ZsYXNoUmVhZGVyP3htbGlkPTk3ODAxMzIzNzU1ODAvMQ==

Next missmatch i found:

D&D Q10

the dump of nagamin , 6. field is blank
the dump of Valerio , 6. field is filled with “sip”

What is the correct statement?

with SIP must correct one.

please

HOW many ip phone calls can be sent a 64-kb/s frame relay link that uses the G.729 codec ?
codec? the sampling rate is 50 times a second, with 20 bytes pero sample. There are 8 bytes of frame relay header overhead with no checksum, and header compression is used.

a–3
b–4
c–5
d–7

????
please .. any help me

I think c: 5calls

The payload is 20 byte + 4 byte (RTP + UDP + IP) as RTP Header Compression and + 8 byte FR.

Then you have 32 byte x 50ms x 8 = 12,8k per call

@nagamin
When will you take your exam dude?
I will take my exam as soon as possible, but
in this moment i see many guys they failed (look to examcollection),
or pointed it exactly like Andren or nearly 800.
I see no clear pass over 850 marks, are there maybe some questions in your dump incorrect?

nope, thats would be my first exam in ccnp voice.

Whats your opinion to this question from a user in examcollection?

default power used by a cisco 802.3af compatible ip phone
options were
6.3
7
15.3
0
———
Standard 802.3af Class = 0 / Maximum Power (W) = 15.40 / Class Description = Default

So my opinion ist 15.3W, you/or anyone else agree this?

On my exam i put 15.3 .

I think Q50 is wrong, Cisco Unified Border Elements to provide network hiding.

Intelligent ip address translation should be a correct answer

http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps5640/product_data_sheet09186a00801da698.html

Correct Answer for Q 50 is:

B. IP network security boundary
C. IP network privacy
E. Intelligent IP address translation for call media and
signaling

You can read this in Section “Key Features of the Cisco Unified Border Element”
on Page 5-51

Whats about Q16?

dumps says Answer D, is this OK?

Couldn’t it be Answer C?

@ NewInVoice
I don’t think it could be C because it says 245 not 225.

@ NewInVoice and other

here is my step on CCNP voice,
1/ 642-447 (CIPT1 V8.0)
2/ 642-457 (CIPT 2 V8.0)
3/ 642-467 (CAAPS V8)
4/ 642-427 (TVOICE V8)
I’m now on 642-437 ( CVOICE V8) for the last. So previous exams studies will help you alot.
642-427 and 642-437 is the most tricky exams in CCNP Voice I learn.
Questions are did not come in strait. Like word by word from cisco guide.
Be prepared.

I think i find the wrong Answers, can anyone agree my opinion?
Did anyone find other errors?

Q1 is not C , answer is A
Q10 put “sip” to the last free box
Q39 place “media flow through” instead of “media flow around”
Q50 is not BCE, answer is BEF

Whats about Q14, some dudes who did the exam says that there
a 5 boxes to place. Is it posssible that “default dial peer” comes in
the first box, or must be only 4 fields used?

for 1 some belive in C.
10, 39 50 what you say is correct.
On 14, if there is “Not all options are used.” was at the end of question.
So, 4 boxes for answer can be right.
If not there, must be 5 boxes. Or there is two questions in question set. ????

For Q1 now i agree with you nagamin:
On Page 6-75 i found the answer C word by word:
“Marking packets or frames sets infonnation in the Layer 2 and Layer 3 headers of a packet so that the packet or frame can be identified and distinguished from other packets or frames.”
So i will swop to C for Q1

For Q14 i think:
– if there 4 Boxes then without “default dial peer”
– if there 5 Boxes then with “default dial peer”

When will you take your exam nagamin ?
I wish i had pass this 4 exams like you did 😉

nagamin im in the same position as you. My CVoice expired so I have only this one left. Lets kick butt. I plan to take this exam next week. can we confirm from the 145 questions pool the first 70 something are correct from previous test takers?

I think up to 80 is good, plus some new questions from test taker comments.
I’m working on it and link will provide soon.

thats great. let me know how I can help. I actually have the new cisco press 642-437 to reference if needed. Send me your email and I can get it to you if you dont already have it

Failed 778.
went with most of what was on the dump and

Q10 put “sip” to the last free box
Q14 put default dial peer in the last box
Q39 place “media flow through” instead of “media flow around”
Q50 BCF

Additional question about DiffServ. Thought it was easy.

hey John, can you please email the new cisco press 642-437 to qkiqt@hotmail.co.uk
Thanks.

Corretion:
For Q.56 ValerioVB is right, must be Configure CoS Level, trusted, and untrusted. Untrusted is default.

For Q 56 I agree with you nagamin

On Page 3-50 you can read this:
– Trusted
– Untrusted (default)
– Configured CoS priority level:

Q.39 “media flow through” Or “media flow around”?
Cisco UBE might either proxy the media channel, which is referred to as “flow-through”,
or let the media channel pass through the gateway without any modification, which is referred to as “flow-around”.
The media proxy function is necessary when the VoIP traffic parameters of the incoming call
leg differ from the VoIP parameters of the outgoing call leg.

In my dump, question said that-
The IP WAN connection between the Cisco Unified Border Elements uses SIP. Not all commands and spaces are used.
So both call leg are same SIP traffic. There is no modificationWe sould use “flow-around”.
Any suggestion?

If without “The IP WAN connection between the Cisco Unified Border Elements uses SIP. Not all commands and spaces are used.” and as shown on exhibition “media flow-through” is correct answer.
Please be aware of signaling protocol.

Please correct me.. I just did the 642-437 exam got 778 just 12 points short…

How does LLO ensure that voice traffic is always expedited?

A. LLO adds a strict priority class to CBWFO. This class allows delay-sensitive data such as voice to be dequeued and sent first.
B. LLO uses CBWFO to prioritize voice traffic and dequeue the voice packets so that they can be handled first.
C. The strict priority queue has a higher weight than the queues in CBWFO. This weight allows the delay-sensitive data such as voice to be dequeued and sent first.
D. The LLO strict priority queue allows delay-sensitive data such as voice to be dequeued and sent first (before packets in other queues are dequeued), giving delay-sensitive data preferential treatment over other traffic.

P4S Ans: D
ItExam and AT Ans: A

Please which one is correct ?

Q5
nagami dumps says ans: A
Valerio dumps says ans: D

Please advice

Q.5 correct answer is D

Low Latency Queuing:LLQ provides strict priority queuing (PQ) in conjunction with CBWFQ. LLQ configures the priority status for a class within CBWFQ, in which voice packets receive priority over all other traffic.
Refrence:Using QoS to Improve Voice Quality

Also on Cisco Site-

Feature Overview:
The Low Latency Queueing feature brings strict priority queueing to Class-Based Weighted Fair Queueing (CBWFQ). Strict priority queueing allows delay-sensitive data such as voice to be dequeued and sent first (before packets in other queues are dequeued), giving delay-sensitive data preferential treatment over other traffic.

Ref:”http://www.cisco.com/en/US/docs/ios/12_0t/12_0t7/feature/guide/pqcbwfq.html”

I took the 642-437 and passed… Many questions from dumps were there. Questions that got me were the areas I should have concentrated on more.
(1) DSCP and COS calculation (will try to remember the context of the question & update later)
(2) Address translation configuration – taking a 3 digit and converting to a 10 digit dial pattern.
symbols /^3….//$\4801234 etc… (not exact but I think you get what I mean)
It seems I got at least 2 or 3 questions that were not shown in the dumps I used. However, I know that is something I will focus on a bit more later.

Q.11 D&D
Nagamin Dumps: TCP-Q931-Q931-Q921-Q921-ISDN Call Control

VallerioVB: ISDN Call Control-TCP-TCP-Q921-Q921-Q931

Which one is Correct Please update….

Thanks in Advance

@Shaik SA

VallerioVB: ISDN Call Control-TCP-TCP-Q921-Q921-Q931

is correct one.
His pdf was latest updated from ActualTests.

Thanks nagamin
I will try the exam this week
once i finished i will share my experience…

Nagamin, which answers did you identify as wrong?

Please read previous members comments. There is about 7 or 8 answers had been corrected.
Also you can find new questions. Do not relied on my dump answer only.

Hey nagamin,
is it possible that you upload a corrected version?
So anyone can compare its own version to yours.

Here you go.
Good Luck Guys.
http://hotfile.com/dl/125618990/e957ec2/642-437_Collection_08-01-2011.vce.html

Here are the default values for the mapping until you decide to change it

CoS-to-DSCP mapping

•CoS 0 => DSCP 0
•CoS 1 => DSCP 8
•CoS 2 => DSCP 16
•CoS 3 => DSCP 24
•CoS 4 => DSCP 32
•CoS 5 => DSCP 40
•CoS 6 => DSCP 48
•CoS 7 => DSCP 56

DSCP-to-CoS mapping

•DSCP 0 => CoS 0
•DSCP 8.10 => CoS 1
•DSCP 16.18 => CoS 2
•DSCP 24.26 => CoS 3
•DSCP 32.34 => CoS 4
•DSCP 40.46 => CoS 5
•DSCP 48 => CoS 6
•DSCP 56 => CoS 7
..

Any one notice that Cisco Exam on Pearson VUE was 200S US today?

nagamin this is really great from you, THANK YOU.

I compared you dump against mine.
There are exactly the same results that i have notice for me.

Only for Q85 you have an error inside.
The answer must be A (E&MType1) not B (E&MType2)

I didnt understand your question:
Any one notice that Cisco Exam on Pearson VUE was 200S US today?

Exam fees is increase to 200$ US per exam on CCNP voice track.

How many questions have the exam?

There is 70 questions and 90 min.

I’m done with CCNP Voice, last dump with 90 question is enough to pass.
But I did not make it 900 over. If you guys not satisfied with under 900 hang on for a while, someone will update corrected one. All question are from last dump, except 3 question.
one was descripton of DiffServ.
Q.14 with default dial peer.
another one was When you place a call and the vocie gw has a pots and voip dialpeer in configuration, what path does the call take

bad % section on
implement cisco unified Border Element
Describe the need to implement QOS for voice and video

Good luck guys

Nice to hear nagamin, CONGRATS to YOU!

Can you tell me what is the answers from the
1. descripton of DiffServ
2. When you place a call and the vocie gw has a pots and voip dialpeer in configuration, what path does the call take

@NewInVoice

1. descripton of DiffServ

DiffServ model has these key benefits:
• It is highly scalable.
• It provides many different levels of quality.

If you remember those two you will be fine. You must select those in exam.

2. When you place a call and the vocie gw has a pots and voip dialpeer in configuration, what path does the call take

I chose VoIP dial peer, unless WAN outage.

I want to push you guys to take the ASAP.
I feel like question gonna change soon. Price is increase, that mean more time will be allow.
I’m not sure.

Crap! After I paid my exam for 200$, VUE exam voucher is come out.

check here.

http://www.getcertified4less.com/product_p/cisco640-642promo.htm?click=19434

@nagamin
So how long did it take you to complete CCNP VOICE?

thanks

I am sorry but in the question :

When a Cisco Unified Border element connects two voIP streams using flow-around media, wich of the following options call that the flow around and the components that flow through the device?

I think the right answer is B:

Call signalling flows through and call media flow around the device

Because the explanation is clear: signalling always through CUBE, Media can flow around or through

am i right?

Hello,

I belive that the are another wrong answer.

Wich three of the following methods are used by Cisco Unified Border Element to provide network hiding (choose three.)

F. intelligent IP address translation for call media and RTP flows, is marked like good

But in key Features :

Intelligent IP address translation for call media and signalling

Call media and RTP flows are the same.

Rigth Answer is : C. media flow-through

Please anybody rigth with me

Can any1 share the valid dump for CVOICE pls 🙂

Into this thread there are two, both very good

Q24 questions from AT 5.2 that Valerio provided if the PSTN DID range in site B is 300-555-1234, the answer will be letter B?

Question 48 should be A, C and D. You wouldn’t use FRF.12 on a Point-to-Point link

72 is B

92, should be E. Type 2 is used for “Geographically Separated” because there is no common ground

132 should be D. The “up” parameter in the config says we are starting at the bottom and are going up.

@madaboutvoip

Network hiding
• IP network privacy and topology hiding
• IP network security boundary
• Intelligent IP address translation for call media and signaling
• Back-to-back user agent, replacing all SIP-embedded IP addressing
• History information based topology hiding and call routing

http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps5640/product_data_sheet09186a00801da698.html

Q.50 must be: B,E

– IP network security boundary
– IP network privacy

In the question there are the possibility F:

– Intelligent IP address translation for call media and RTP flows

but the right one is:

– Intelligent IP address translation for call media and signaling

In the question there are the possibility A:

– Back-to-back user agent, replacing all H.323-embedded IP addressing

but the right one is:

-Back-to-back user agent, replacing all SIP-embedded IP addressing

What is the right 3rd answer???

CIPT1 v8.0 642-447….
any one has this paper’s dumps

Took it and failed. restudying and just found this site. I’m trying to go through an verify every answer on AT5.2.

I’m also reading reading version 4 of the cvoice book and comparing the answers to what is written in the book.

as for for question 1. It is C. Direct sentence out of the book on page 615. A is not the answer because CoS is not a layer 3 field.

@Taurus the correct answer has to be B, C, and D. they changed a word in A nd F and made them not right. Here is a link to a cisco faq where it calls media flow-through network hiding :

“Media packets can either flow through (thus hiding the networks from each other) or around the Cisco Unified Border Element platform.”

http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps5640/prod_qas09186a00801da69b.html

Cisco making the answers this tricky really ticks me off. Ok that’s my only vent.

AT5.2

Q8 should be B. IP RTP priority and Frame Relay IP RTP priority work with WFQ and CBWFQ.

page 283 CVOICE book

that should be page 263

@whuma
I think answer D is right at Q. 8

http://www.cisco.com/en/US/docs/ios/12_0t/12_0t7/feature/guide/pqcbwfq.html

LLQ also works with CBWFQ and WFQ
That can you find at the benefits.

@taurus

I saw the same thing but it just says it is a related feature. it does not explicitly state that LLQ works with WFQ.

When a answer to a question is seems like it could be either, I am going to go with the answer that is a direct sentence from the CVOICE book. The CVOICE book does not mention LLQ working with WFQ it only states working with CBWFQ.

P 263:

IPRTPPriorityandFrameRelayIPRTPPriority:Providesastrictpriorityqueuingschemethatallowsdelay-sensitivedata,suchasvoice,tobedequeuedandsentbeforepacketswhenotherqueuesaredequeued.Thesefeaturesareespeciallyusefulonslow-speedWANlinks,includingFrameRelay,MultilinkPPP[MLP],andT1ATMlinks.Itworkswithweightedfairqueuing(WFQ)andclass-basedWFQ(CBWFQ).