Share your CVoice v8.0 Experience

for the 802.3af question I believe the correct answer is 6.3W. The other values do not match the class value of af. All of Cisco 802.3af compatible phones come up in a low power mode so that they are compatible with the pre-standard POE.

from Cisco white paper:
http://www.cisco.com/warp/public/cc/so/neso/bbssp/poeie_wp.pdf

A Cisco IEEE+CDP powered device, such as a Cisco IP phone 7970G, will come up in low power mode (6.3W) and
will transmit a Cisco Discovery Protocol message with an inline power (ILP) type length value (TLV) that informs
the PSE of the actual power required by the device

If the requesting powered
device exceeds the power budget for the line card or switch, the port will be either powered down, or the port will
remain in low power mode (7W).
This management scheme is implemented to provide backward compatibility and investment protection to the
installed base of Cisco Catalyst prestandard Power over Ethernet capable line cards and switches. Cisco IP phones
are power efficient and require 6.3W maximum power as reflected within the prestandard Power over Ethernet
implementation. However, the development of new high-power powered devices, such as wireless access points and
IP phones with color LCD screens require additional power that cannot be delivered using the prestandard
implementations. By bringing up Cisco powered devices in low power mode, Cisco high power powered devices can
operate, albeit with reduced functionality,2 on prestandard line card

The right answer to 802.3af is 15.4w, but i find other version of The question, don’t speak about 802.3af but Power in line ( Cisco pre-802.3af), in that case The rigth anwer is 6.3w.

i pased The exam, today 🙂

@madaboutip

I didn’t remember 15.4 being a choice. I thought it said 15.3W

did you get the question about the voip and pots dialpeers where the destination pattern matched and which route would the call take (IP WAN or PSTN both are operational).

what do you think the answer is ? PSTN, IP WAN or random (if it was a choice)

also I remember another question about CUCME. something to the effect that all existing phones on the CUCM are working but you are adding another one and it is not working.

Guys – great info! I have to agree with whuma on the power question due to the whitepaper and the fact that the question has 15.3W as an answer. If it were not for the 15.3W instead of 15.4W it would be more of a dilemma. madaboutip did you specifically see 15.4W as an answer? Are you certain that you got that one correct?

Which two functions are associated with a voice gateway? – all of the answers suck! Any thoughts?

@scooby. I’ll give you my opinion on functions of a voice gateway.

I believe the answers are A and C.

From the CVOICE book p.6:

Depending on the deployment type, a gateway can perform one or sever-al of these functions:

■Act as a voice switch that interconnects multiple traditional telephony circuits. Thecircuits can be analog or digital. The gateway participates in signaling and might haveto convert the media channels. Gateways provide physical access for local analog anddigital voice devices such as telephones, fax machines, key sets, and PBXs.

■Act as a PSTN-to-VoIP gateway that provides translation between VoIP and non-VoIP networks, such as the PSTN. In addition to the functionality of traditionalvoice switches, the PSTN-to-IP gateways enable voice and video communicationsbetween traditional PSTN infrastructure and converged IP networks.

■Act as a Cisco Unified Border Element (often written as Cisco UBEor CUBE) that in-terconnects two IP networks and allows communications between endpoints distrib-uted among them. The Cisco UBEs might implement filtering, address translation, andsecurity-related functions.

answer A. clearly matches the first function.

answers D and E are functions of H.245.

answer B just doesn’t seem right.

answer C closely matches what a CUBE does: in-terconnects two IP networks and allows communications between

I agree with your reasoning – the question is a POS for sure! Thanks for your answer. As usual these questions have more than a bit of ambiguity to them.

Anybody – Q34 about the small office that needs to provide outbound dialing and inbound DID. All signaling is loop start. At first examination I would say “A” however DID function does not support loop start. If that loop start was not there. I am not aware of being able to DID without either a digital interface or using a FXS/DID interface. Thoughts?

scooby, I’m with you. I’ve been struggling with that one because I saw the same thing you did about loopstart not support on FSX-DID cards. The only thing I could come up with is that it is an example of one stage dialing, where the direct inward dial is configured in the dial-peer and some the config has to be missing (the part that assigns the phone number to a voice port). the other configs have commands that don’t exist or are not formatted properly.

Yep..this may be a wait and see what the exact question is on the test..armed with the knowledge that loop start is not supported on FXS-DID. I am no certain that “B” is the answer. As far as I know the only way to get inbound DID is with a FXS-DID interface or digital. Perhaps two stage dialing where the PSTN caller gets routed to an IVR script or an AA and then inputs digits – that could work. But is the configuration shown in “B” even valid with direct inward dial command on the FXO interface?

Thoughts on Q43? Could be either “the class map will not map the packet and no QOS will be applied” or “for the packet to be forwarded to the policy map, it must have a ~”

it specificially asks what the call map will do if a packet arrives with cos of 6 and DSCP value of EF.

Therefore to me the answer is it will not map the packet and no QoS will be applied.

I don’t like the work “map” in the answer because a class-map matches or doesn’t match.

Answer D would better fit the question when will a packet be matched by the class-map above.

Which two functions are associated with voice gateway ?
correct answers:
A) Swithes voice channels between connected analog and digital voice circuits
C) Interconnects two logically separate VoIP networks

Q.55 in which situation would the trust boundary be located at the access layer.

A if the endppoints, both IP phones and PCs, are incapable of marking traffic properl
B. if PCs are switched through IP phones and the IP phones traffic can be trusted to mark both traffic streams properly.

from CVOICE page 617

If the end device is capableof performing this function, the trust boundary for the network is at the end device. If the device is not capable of performing this function, or the access layer switch (for example, a wiring closet switch) does not trust the classification that is done by the enddevice, the trust boundary might shift.

How this shift happens depends on the capabilities of the switch in the access layer. If theswitch can reclassify the packets, the trust boundary is in the access layer

after reading that A seems to fit better to me.

thoughts?

@whuma. About Q.55 i am agree with you.

@whuma and Azon – crap I spent 20 min earlier looking into that one. whuma your find in CVOICE really made me question my previous findings and look again. You have a valid point however I think we need to consider that if the phone has a switch in it and if it can be trusted to mark both streams properly then it too becomes part of the access layer and not the end device as the PC is actually the end device. If we consider the command MLS extend COS are we really extending the access layer perhaps??? SOB I hate this ambiguity that is built into these tests!

@whuma and Azon – Here is a link to the campus QOS design guide. The definition that it provides for trust boundary is as follows: The primary function of access edge policies is to establish and enforce trust boundaries. A trust boundary is the point within the network where markings such as CoS or DSCP begin to be accepted. Previously-set markings are overridden as required at the trust boundary.
You should enforce trust boundaries as close to the endpoints as technically and administratively possible as shown in Figure 2-3.

If we take that definition of the trust boundary and apply it to the question “B” becomes the correct answer.

http://www.cisco.com/en/US/docs/solutions/Enterprise/WAN_and_MAN/QoS_SRND/QoSDesign.html#wp998284

Cheers – it is 12:30 (midnight) and I have to get to bed!

the anser is shown as C on Q63 but the ip address command is not correct. is this a mistype?

Q64 makes no sense to me.

Q71. four types of ephone-dn supported by SCCP in CUCME

A,B, D are easy to see.

C is SIP only

D and F . Both are supported by SCCP which leaves 5 correct answers for four choices.

What am I missing ?

@ whuma
Q.63: Why is the ip adrress comand not right? I think this is the true answer – C.
The ip adrress of the zone local HQ and the id of the zone BR (the own zone)!

Why make Q.64 no sense? Assume the NANP numbering plan!

Q.71: You´re right. There are 5 possible answers – but choose the four best. That´s Cisco 😉

@taurus

the command syntax is “ipaddr” not “ip address”

Q64. it says all calls to the PSTN

Q.63 is just confusing to me. the AT5.2 explanation makes no sense.

all calls to the pstn are routed through site B. if users sometimes dial 4 digits to get to b then why translate to 10. ?

Q.81. Answer is A not C.

http://www.cisco.com/application/pdf/paws/14368/plar_config.pdf

Connection trunk mode is a permanent connection. The VoIP call is always connected independently
of the plain old telephone service (POTS) port being on−hook or off−hook.
·
Connection PLAR mode is a switched VoIP call. The call is setup on an as−needed basis. With
connection PLAR, no bandwidth is consumed while the phone is on hook. When a phone connected
to a POTS dial peer is taken off−hook, the call is automatically connected and the remote phone
begins to ring.

I was wrong on Q.81 it is C.

Although a tie-line connection is similar to a trunk connection, it is automatically set up for each call and torn down when the call ends.

***Need input taking the test soon******

I’ve have tried my hardest to find configuration examples to verify the cube implemenation scenarios and I’ve found bits and pieces but no true way of know what the answer is. Anyway:

Q10. why add the SIP command ? without any additional configuration what does it do ?

Q11. I found a diagram on this web page matching exactly except the exam is missing an oval. so I think it should be Q.931 tcp tcp q921 q921 q931

http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a00801da84e.shtml

on Q39. Why even configure the codec transparent command on both dialpeers and the media flow-through command on the one dial-peer. media flow-through is the default operation and is only the allowable configuration for h323 to SIP

Q103. AB and C or F? F SIP cause codes are directly supported between H323 and SIP. the IPSEC to TLS explanation doesn’t seem to fit as well.

Q141. B or D ? do you use ACLS to classify voice control traffic or use ACLS to mark voice traffic. the voice traffic should already be marked by the IP phones.

thoughts appreciated.

@whuma
Q.141: It´s answer D. The ACL only classify the traffic , they don´t mark the traffic!

@taurus

Q141. I understand but the only difference is those two answers is the second line.

B. Use ACLS to classify voice control traffic
or
D. Use ACLs to classify voice traffic.

so the question is in a LAN do you classify the voice traffic or classify the voice control traffic.

the voice traffic is already marked by the IP phone. The voice control traffic I’m not sure is always marked.

@whuma
I think in the first step voice is more important than signaling ???!!!

@taurus

I agree that’s what I’m going to go with is D

For your information, you will not be see the question which is greater than question 80 in real exam.
Don’t ask me why. Time and tide wait for no man.
Good luck.

@MadMax: I don´t understand your statement?! Which question 80? could you explain it, please.

In CVoice dump, you guys need to concentrate on Q.1 to Q.80. The rest is extra or from old exam. But I’m not responsible for. 95% is from 1 to 80. Total of 65 to 70 question.

can anybody help me with the proper dumps for CVoice 642-437

What´s going on with Q. 72: When a Cisco Unified Border Element connects two VoIP streams using flow-around media, which
of the following options call that the flow around and the components that flow through the device?

I think answer B is correct and not C:
B. Call signaling flows through and call media flows around the device.

Any thoughts…

Signalization always go through CUBE, media flow depending on the mode. So correct answer B. Call signaling flows through and call media flows around the device

Guys, I just took the exam and was shocked to see my score 766. Thus, I need to retake.
FYIThe mistake margin on this exam is extremely narrow!

I am a bit convern about the answers though specialy drag and drop when configuring CUBE and dial peer for the CUBE. The questions that I am concern about are as follows:

1) Drag the appropriate IOS Command to configure the dial peers for the CUBE (Cisco Unified Border Element). The IP WAN connection between the two CUBE uses SIP. The commands according to the dump are as follows:

dial-peer voice 1000 voip
destination-pattern 1…
sessiontarget ipv4: 10.1.1.1
codec transparent
media flow-through
!
dial-peer voice 4000 voip
destination-pattern 4….
session target ipv4:10.1.130.1
codec transparent
session protocol SIPv2.

Guys any input on this one?

Q.2) Next one is similar but you are simply configuring CUBE:

voice service voip
allow-connections h.323 to sip
allow-connnections sip to h.323
h.323
call-interwork
sip

Does the configuration ends at “allow-connections sip to h.323 or do we need to include the “interwork” command since the question doesn’t ask for any additional configuration. The first 3 lines simply configure the CUBE to allow the traffice between sip and h.323 and vice versa.

Any input will be greatly appreciated!

Q3) Processing delay? Is it fixed or Variable?

I chose fixed on the exam!

Q4) Does ” Call media flow Through or Around”? in the CUBE question?

I chose the option:

Call media flow through and call signaling flow around

Please help me since I am planning on retaking the exam soon!

@Ali
Q3: Processing delay Is Variable!

Q4: Signalization always go through CUBE, media flow depending on the mode.

Q1 + Q2: What is here your Question?

Q1 is right – I think so!

There are new Questions in the exam? How many? Do you know some of it?

@ taturus
One of the new question I remember is about Diffserve:

Q.1) What are the benefits of Diffserve?

A) SCALABLE
B) Can be used in large number of class services
C) based on flow Architecture
D) Uses flow model to provide QOS

I chose: A and B since Differve is Scalable and supports large # of classes. I feel pretty confident about this answer. However, couple of drag and drop to configure the CUBE are bit confusing according to the answer in the dump.
Q2: “configuing CUBE”. I am pretty sure about the following commands:

voice service voip
allow-connections h.323 to sip
allow-connnections sip to h.323

However, in the dump it also adds the following commands:

h.323
call-interwork
sip

Can you confirm the above?

In addition, please confirm ” session protocol SIPv2.” command in Q1. Does it need to be included?

@Whuma

Can you or others please confirm that

“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”

is it true? I agree with you, CoS is not a layer 3 field, but i’d like to be sure about that

thank you

@Ali:
Q2: you need the 3 additional commands like:
h.323
call-interwork
sip

Q1: I Think you need all commands, but I´m not sure.

How many new question there are in the exam?

@ taurus

The helpful answer to your question is what questions are there instead of ‘howmany’. As far as I remember:

1) Desribe the TWO benefits of Diffserve
2) Given the dial-peer config output and exibit with phones connected through the gateway into the WAN and PSTN Cloud. Questions asks which path does the packet takes?
3) Steps when IP Phne registration
4) Variation of the “PSTN DID range” question with the exibit.

Just do a bit studying and I think you should be fine. I am taking the exam next week.

Hope this helps!

What about “session protocol SIPv2” command in Q1? Why do is it needed? I am researching on this. If you find something please let me know.

How can this be correct? There are two questions, same answers, but in one the last option of default dial peer isn’t used.

“The voice gateway selects an inbound VoIP dial peer by matching the information elements in the message with the dial-peer attributes. From the list on the left, drag the elements to the right and drop them in order in witch a voice gateway matches inbound calls.”

“The voice gateway selects an inbound VoIP dial peer by matching the information elements in the message with dial-peer attributes. From the list on the left, drag the elements to the right and drop them in the order in witch a voice gateway matches inbound calls. not all options are used.”

From the CVOICE 8.0 book:

“Inbound dial-peer matching: Performed with these commands, in this order: incoming called-number, answer-address, destination-pattern, and port. If no inbound dial peer is matched, the default peer is tried.”

So one of these are wrong? I’m guessing it’s the one that says you don’t use the default dial peer? Any thoughts?

Thanks

@ ciscodude:

The answer is correct:
incoming called-number, answer-address, destination-pattern, and port.

Default option is only used if there is no criteria specified.

Hope that helps!

Guys, The Q1 in the test. The correct answer is ‘A’ not ‘C’!

@Ali:
Why is the correct answer for Q 1 answer A??? Why Do you knows this?

@all:
Q 10: Drag & Drop
Why i have to use the commands “323”, “call start interwork” and “sip”??
I think i only need:
voice service voip
allow-connections h.323 to sip
allow-connnections sip to h.323

The command call start interwork is for h323 -to-h323 interwork. Any thoughs….

The same at Q39:
I think we only need:

dial-peer voice 1000 voip
destination-pattern 1…
sessiontarget ipv4: 10.1.1.1
!
dial-peer voice 4000 voip
destination-pattern 4….
session target ipv4:10.1.130.1
session protocol SIPv2

Because of: There is no question for codec transparent and media flow-through is default.
Any HELP….

thanks!

@taurus
Q39: We do need “codec tranparent” command otherwise we have to specify the type of codec (G.711, G.729 etc). However, I do agree on the “media flow through” that is default. Hence, doesn’t need to be entered.

Note: The questions on this exam apear randomly so my apology for not being particular. However, there is a question to configure the GW with the GK that is not correct!

@ All
I agree with taurus on the “call interwork” command that disables the communication between H.323 to any or vice versa so no need for it and H323 and sip is not needed since it is not asked!

@Ali: I don´t understand your explanation of question 39. Why we need codec transparent??? When you look on question 10 on the given config: there is no codec transparent too?

Any opinion?

Got another one here I think might be wrong. Check it out, let me know what you guys think.

Drag the appropriate IOS command from the left and drop them in the spaces on the rgiht in order to configure CUBE. The ITSP does not support early offer.

The answer listed is:
voice service voip
allow-connections sip to h323
allow-connections h323 to sip
h323
call start interwork
sip

————————————
I believe the sip at the bottom is not needed, anyone have thoughts on that? When you type h323, that’s only bringing you to the h323 configuration right? It’s not an actual configuration command? The same goes for SIP right?

@taurus

I would think it would be this for Q39:

dial-peer voice 1000 voip
destination-pattern 1…
codec transparent
sessiontarget ipv4: 10.1.1.1
!
dial-peer voice 4000 voip
destination-pattern 4….
codec transparent
session protocol SIPv2
session target ipv4:10.1.130.1
———————————————-

We are crossing the WAN, so flow through is ideal, which is default so no command needed. Session target I believe goes last, at least that’s what I see in the CVOICE book. I would say the codec transparent is needed, if it doesn’t specify to use one, I’d say let the other devices decide right? So tell the CUBE to be transparent in that respect.

As far as the other question goes, I think you are right, call start interwork should not be needed. From what I can find that is only used on H323 to H323, so the answer would be I believe:
voice service voip
allow-connections sip to h323
allow-connections h323 to sip

@ciscodude:

@ciscodude:

I totally agree with you!

@Anonymous

I agree with you in regards to the “codec transparent”. If codec transparent command is not used then we have to specify the codec which consume more DSP resources, since the question doesn’t mention to use a particular codec then codec-transparent command will allow the CUBE to leave the codec negotiation upto the endpoints.

Some resources I have found:

CVOICE p,551:

“the Cisco UBE is configured for a transparent codec and will
leave the codec information contained within the call signaling untouched”.

Another source:

Q. What is a transparent codec, and what does it do?

A. The Cisco Unified Border Element transparently passes capabilities between endpoints. To configure this function in Cisco IOS Software, a new codec type called the transparent codec is used.

The transparent codec is unique to the Cisco Unified Border Element. Configuring codec transparent on the Cisco Unified Border Element allows it to pass through codecs that it understands, but it does not force the negotiation of any particular codec-codec negotiation is left to the two endpoints. Only codecs that are supported on the Cisco Unified Border Element can be passed between the two call legs.

Hope that helps!

Again, I think this is wrong. Someone else want to weigh in here? Also is it just me or is this question worded poorly.

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

A) All security information flows through the CUBE, and all call signaling and RTP flows around the device.
B) Call signaling flows through and call media flows around the device.
C) Call media flows through and call signaling flows around the device.
D) The initial call-signaling traffic flows through the device to intiate the call, and all subsequent calls flow around the device.

Now they answer given is C, clearly that has to be wrong? With Media flow-around, call media flows around, not through. Call signaling is what goes through.

From the CVOICE 8.0 book:
When using media flow-around, CUBE leaves the IP addresses used for the media connections untouched. Call signaling will still be processed by CUBE, but after the call is set up, CUBE is no longer involved with the traffic flow.

So to me, the answer clearly is B, call signaling flows through, call media flows around.

Sound right to you guys?

How to open all the 145 questions inside file ?
Cisco.Pass4Sure.642-437.v2011-07-19.by.nagamin.145q.vce.html

@ciscodude

I agree with you, there’s no way for call media flow to pass through the CUBE while signaling flows around, it’s the opposite. Answer B is the correct one

What about Q.64? I still don’t understand if the correct answer is A or D, what do you think?

My questions must be numbered differently. What question are you referring to? I can look it up in the book.

On the SIP drop down, it specifies only the signaling information. So, do you put the RTP stream in the last box or not?

Also on Q.91, I’m still not sure that the 4-digits is the right answer. The diagram shows the Merrimack IP phone having a 10-digit number assigned. Doesn’t it make more sense to strip the 6 and add 512555 before sending it on?

Q1. TOS is a Layer 3 packet header and COS is a Layer 2 frame header. Shouldn’t the answer be C?

Q11. Is the UCM termination Q.931 or ISDN Call Control? The diagram shows Q.931.

Q.56 seems like it should be CDF, not ACD. switchport priority-extend doesn’t have an option to trust dot1q, just cos and trust.
http://books.google.com/booksid=_hGLWTaBk0gC&lpg=PT457&ots=EfK7nY0J4L&dq=instructed%20to%20treat%20the%20Layer%202%20CoS%20priority&pg=PT457#v=onepage&q&f=false

Q.59 According to the explanation, the answer should be Distribution of dial plan logic, conforms to international and private standards, simplicity in provisioning, reduces post-dial delay, and availability and fault tolerance.

Sorry, link is
http://books.google.com/books?id=_hGLWTaBk0gC&lpg=PT457&ots=EfK7nY0J4L&dq=instructed%20to%20treat%20the%20Layer%202%20CoS%20priority&pg=PT670#v=onepage&q&f=false

@ciscodude, I’m referring to Q. 64 of AT5.2, that is:

Site A uses three-digit internal numbers and remote Site B uses four-digit internal numbers. All
calls to the PSTN are routed through Site B. What dial plan below best represents provision
simplicity, assuming the NANP numbering plan?

A. Translate all called numbers at either site to ten digits.
B. Translate all called numbers within Site B to three digits.
C. Translate all called numbers within Site A to four digits.
D. Translate all called numbers leaving Site A to ten digits.

by the way I took the exam today, passed, score 836 😉

Good luck to everybody!

So, did you answer A or D, @Davi81?

I´ve passed today!
I reached 100% in the “Implaement Cisco Unified Border Element” section!!!!

I answered:

Q 10: Drag & Drop
voice service voip
allow-connections h.323 to sip
allow-connnections sip to h.323

Q39:
We only need:

dial-peer voice 1000 voip
destination-pattern 1…
sessiontarget ipv4: 10.1.1.1
!
dial-peer voice 4000 voip
destination-pattern 4….
session protocol SIPv2
session target ipv4:10.1.130.1

That´s all!!!!!

There are the new questions mentioned in this forum. One like:
“… the challenge to implement Variable Length Dial Plan”
There´re five possible answer – do you have to choose one.

Another new question about CoS and DSCP Mapping in a Switch when this is configured and a picture with a DSCP-to-Cos Mapping Map:

#mls qos trust device cisco-phone
#mls qos cos

And a new question when you have the binary IP Precedence 011: What is the binary DSCP

and another one, but I´dont remenber.

Good luck

Thank you!

@taurus, awesome man. Thanks for the information!

@ Taurus

Congratulations on your success! You deserve 100% on the CUBE section. So, when are you planning on studying for the CIPT1 exam? We can possibly target it together.

I has two new question :
1) Description of call legs (interconnect physical end-device; maps ip address + 3 options )
2) In which mode RSVP CAC functionate (Media-flow through only; Media-flow around only + 2 options)

@ Azon

1) Can you remember the other 3 options? These two are not the answer.

2) The answer is Media-flow through only. Media flow-around is not supported with RSVP-based CAC.(Pg. 552 from CVOICE Book 4th. edition)

1) Establishes a connection between input and output legs…i sets this variant in my exam
2) Yes, only Media-flow through supported (I chose this variant), but the other two options include an extended formulation of this answer.

Question about registering ephone & ephone-dn, where ephone was registered succesfully but not the ephone-dn. what should be done to correct the problem:
1. Restart Router
2. Restart telephony-services
3. Remove ephone + re-add ephone-dn
4. No create cnf files, create cnf files under the telephony-service

D&D about Fixed and Variable Delay had some differences with the dump:
Proccesing delay instead Packetization delay (and only 2 options for Variable delay).

p/s That is all that i can remember right now

I’ll take the exam tomorrow and will appreciate any comment on few questions from Actual Test v.5.2.

Q.48 They have chosen A, B, D. But maybe B (FRF.12) is wrong because the question refers to
point-to-point IP WAN and FRF.12 works on frame-relay links.
But on the other hand, by ‘IP WAN links’ maybe they mean WAN links that carry IP traffic
and then B is correct. Any comment on this?

Q.50 C (‘media flow-through’) is also right. F says ‘RTP flows’ instead of ‘signalling’ and hence it
could be wrong. But RTP flows = call media, which makes F correct.

Q.56 I think it should be F instead of A. Answer C (‘configured CoS level’), which is also true
means that the IP phone does not trust the CoS of the attached PC and will replace it.
Hence F (‘untrusted’) is also true. Could you comment on this?

Q.61 The explanation says that PBX is using E&M Type V and hence 4 wire operations should be
configured. But I have seen other examples where 2 wire operations is used with E&M
Type V. So, Answer A could be also correct.

Q48. Frame Relay = Layer 2. IP = Layer 3. You can have FRF.12 over a Frame Relay link on an IP WAN.

Passed today with an 813. Better then nothing I suppose. Also passed CAPPS with a 953. On to CIPT1 & 2. Good luck on CVOICE everyone.

@ciscodude Great Job! How much of the dumps and these comments helped you pass the CVOICE with an 813?

for D&D Q about PRI on MGCP gateway
Correct answer:
Q931-TCP-TCP-Q.921-Q.921-Q.931

(CIPT1) v8.0 part1 p. 4-16

Please any one share the dumps, I have purchased pass4sure.. not it’s giving some new questions..
Thanks.

I was able to take the test today and passed with a 835 there were new questions that have already been mentioned in this post I noticed two questions have not been mentioned

1. know the dhcp excluded command it ask you about a question excluding and ip address
2. on q.52 on AT 5.2 its the same question and same answer but it doesn’t have prefix 5

I used the 90q dump and also another dump from testkiller
the official cvoice book 4th ed

Hi,

Please if anybody has CCNP voice V8 studying materials share with us,your help will be appreciated.

my mail is :m.osama_77@hotmail.com

Thanks,

Is somebody able to answer me if the 642-437 exam has labs in it? or everything is drag-drop and multiple choice. Thanks

No lab, drag-drop and multiple choices.

@nagamin

Thanks, I’ve been practicing with the dumps posted here 642-437 Collection 08-01-2011.vce and Cisco.Pass4Sure.642-437.v2011-07-19.by.nagamin.145q.vce do you think those are the latest? In your opinion how much of those is actually in the test

@Hman

use the dump that contains 90q and that should be enough to pass along with the new questions that have been already been stated.

Just wanted to share with you guys how my exam went before I forget ; )
There are indeed 3 questions that do not appear in the dump,
– How much a 8023af telephone consumes, there wasnt an 15.4 watts option so I went with the 15.3
– The same question that already someone mentioned about network delay, jitter delay, dont’ remember the exact options but where something like 20 ms or 30 ms for jitter, network delay up to 150ms another option had up to 400ms

There were 70 questions and if you get a good score on the dump that contains 90q you probably will pass

Thanks and good luck

@ Hman : How much did you score in the exam???

Guys is 90q enough to clear the exam or we need to prepare both 90q and 145q dumps .. Please let me know, because il be writing it in 2 days .. thanks 🙂

@ Aks
I scored 836
I think with the 90q should be enough but you may wanna complete the 145q just so you get an idea of the different types of questions

@ Hman :

Thanks for the Info 🙂

Congrats Hman

Just to my understand, what is the 90Q dump?, I have the latets P4S with 149questions.

Thanks so muchfor your repply.

Congrats Hman

Just to my understand, what is the 90Q dump?, I have the latets P4S with 149questions.

Thanks so much for your repply.

Guys,

Could you please let me know about the 90Q dump? I really appreciate your comments. i have the last version of P4S with 149 questions and would like to understand if is enough to 642-437 exam.

Thank you!

Thanks guys, now I have the 90dumps

hi guys , I have am CCNA(data) certified.
It is possible for me to directly give Cvoice and be CCNA(voice) certified.

Please reply.

@Anonymous October 6th, 2011

No, according to Cisco you need CCNA Voice now.

CCNP Voice Prerequisites

CCNA Voice or any CCIE Certification can act as a pre-requisite.

No it has changed, now pre-requisities for any paper for CCNP voice is CCNA voice.

Guys .. please tell me if 90Q and 149Q dumps are still valid or exam questions has changed?
I am planning to give exam on 14th Oct…
Please reply.. Thanks..

Which best defines an ACD?
I think its answer should be either A or C..

A. a local company that provides phone capability and distribution from the phone company’s
central office
B. a telephone system that is connected to the exchange to provide conventional voice services to several subscribers
C. a telephone system that switches calls between users on local lines
D. a telephone system that responds to a caller with a voice menu and helps to appropriately
connect the call

Any thought?

Q8 is B
From Cvoice Book:
“IP RTP Priority and Frame Relay IP RTP Priority: Provides a strict priority queuing
scheme that allows delay-sensitive data, such as voice, to be dequeued and sent
before packets when other queues are dequeued. These features are especially useful
on slow-speed WAN links, including Frame Relay, Multilink PPP [MLP], and T1 ATM
links. It works with weighted fair queuing (WFQ) and class-based WFQ (CBWFQ).”