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67Q
790 Passing Score
Used 90Q+19Q+86Q+Guide 4th edition. But at most 90&19Q. Dumps are still more than valid to pass. Guide is good to improve the knwlg base.
You only used this dump or some other?
choose three.
I assume A C F
but dump seems right C D F
what is the function of class-based marking?
when a cube connects two voip streams using flow-around media, which of the following options call that flow around.........................
in dump is C right, but I am sure it is B
your Thoughts
Thanks Examcollection!! You did it again!!
this dump along wid 19Q dump is totally valid.....no new questions apart frm these dumps...
NOTE: read statements carefully while answerin d questions as couple of wordings in d options r slightly modified..
i wish all b best to d others
Pass today with 885. The questions were from this dump. Dump still valid
pls can anyone tell me if there is any
simulation for the exam?
Reply pls. Tnx
On score report:
Describe a dial plan 100%
Describe the basic operation and components involved in a VoIP call 100%
Implement Cisco Unified Communications Manager Express to support endpoints using CLI 100%
Describe the components of a gateway 100%
Implement a gateway 83%
Implement Cisco Unified Border Element 50%
Describe the need to implement QoS for voice and video 86%
Describe and configure the DiffServ QoS model 82%
The bottom line is: you can totally trust the dumps. No intelligence is needed to pass.
Thanks to everyone who contributed to creating the dumps. Hope to take my part in the future.
http://softarchive.net/blogs/tuanthanh8889/ccnp_voice_exams_student_guides.776310.html
Q how many call for FR link 128 Kb/s with 50 % data 50 % voice
there is new question for QOS
thanks for all about this kindly support
But confused in few questions of drag and drop and multiple select.
In VOIP drag and drop for dial peer match not port option default dial peer must select.
For no of voice calls question
20 Bytes is voice payload
2 Bytes compressed header
8 Bytes for framerelay
BW per call=(Voice payload + L4+ L3+L2)*Packet ratio*8
BW per call=(20+2+8)*50*8=12000=12KB/S
No of call for 64K BW=64/12=5 Calls
I use for exam preparation: Voicetut, Exam-collection, Actual Test, CVoice 8.0 Book and Voice bootcamp.
Special Thanks to, Voicetut team; Exam collection and all guys here wrote the Comments.
link to voicetut:
http://voicetut.com/cvoice-v8-0-642-437/share-your-cvoice-v8-0-experience
link to a dump is based on voicetut forum:
http://www.examcollection.com/cisco/Cisco.PracticalExam.642-437.v2012-02-15.by.guest77.19q.vce.file.html
Dump still valid. + about 15-20 new questions from voicetut.
Thanks all.
about some question not from the dump.
about 10 question not from the dump.
be careful with prefix
I think the answer of question of "what's the most popular signaling type in North America " is wrong. in dump it's "type 1". but in the cvoice quick reference stated as "type 2".
wish the best for all
thanks exam collection
about 4-5 question not from the dump.
Study hard the previous comments than can make you pass.
this dump is valid.
Thank you all
attention, question 24, appeared tow times, read it very well
question 34, right answer option A not B
question 50, appeared tom times, the diference are in option A SIP / H323, very confused, read it very well
another question about interfces imediat, wink start and delay start, fxs, fxo e&m, confusion PULSE and DTMF
other question tow dial-peers pots and voip sam thestination, who router do to route the call
forget the 145 exam questions, nothing came out
everything else is here
tanks every body,
The answer in the dump is actually correct. 15.4
http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps379/white_paper_c11-481292.html
Very easy to find using google.
When 5551234 is being matched with the outgoing dial peer that is shown in the exhibit, which of the following called numbers will be sent to the PSTN? Exhibit: A. 5551234 B. 1234 C. 555 D. Null E. 5 F. 51234
from the exhibit the prefix 5 command would be missing & hence the ans would become 1234......I also had this original question in my exam.
A: 6.3
B: 7
C: 15.4
D: 0
its asking about IP Phone and answer should be A but in dumps it says C which is used by switch.
Please Comment.
Score 877. This dump is very good, is enough to pass. But no wait for high score.
About 10 new issues. Some have described here and others not.
Also heard from Jan'12 onwards exam fees is going to be increased. Right now it is 200$ US.
5 or 6 question out of this dumps.
many congrats :)
what about the wrong ansowers on this dump , did u changed any ansowers or did the same ansowers?
One question which i remember being out of this-
Voip and Dial peers are -
1) provided destination address
2) provids dial plan for gateway
3)Address endpints of the gateway
IN RSVP CAC for a CUBE which type of media flow method can be used -
1 Media flow through
2) Media flow around
3) Both.
4) ....
New questions = 8
Thank you very much to Nagamin and thanks a lot to Karam and to joy and to
all of you. Dump is still Valid.
open vce file - exam - options - unchecked te randomize options
@Nagamin: brother this fie is password protected so I am not able to convert to .pdf can you please share the password of this file..
Thanking in advance..... Please h
I got this question twice in the exam, but in two forms:
first one as in the dump.
second time with the following details :
dial-peer voice 123 pots
incoming called-number
!
dial-peer voice 2123 pots
answer-address 4087071222
direct inward dial
port 0/0/0:23
a. 123, because of the called-number has pecednce than answer add
b. 213, bcoz of answer address
c.although incoming called has precedence, there is not configured DID ,selct 123
d.although incoming called has precedence, there is not configured port ,selct 123
in this dump some questions have wrong ans.
Q.8 correct ans is B
Q.27 correct ans is A & C
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
Q.10 and 39 didn't asked for d exam
some new questions
1.works with wfq and cbwfq?
a.LLQ
b.ip rtp and priority and framelay...
c.RTP
d frf.12
ANS. B
2.strict priroity queuing and delay sensitive data with cbwfq
a.LLQ
b.ip rtp and priority and framelay...
c.RTP
d frf.12
Ans is A
3. about firmware image file
a........indigual phone .loads file
b........bundile of .load file
c.........cnf file
d..........sbn file
my ans is .cnf (i am not sure)
4. about the dial peer in voice gateway?
5.add new ephone-dn and ephone in cucme. but the ephone is not working. but the other ephone are working normaly those added earlier.then what will do the engineer first?
a. remove ephone-dn
b. remoce ephone and readd ephone
c. restart router
d.no ceate cnf file and create cnf files
e. no telephony service and telephony service
my ans. b (i am not sure)
6.what uses winkstart, loopstart and immediate start followed by pulse or dtmf for call setup?
a.fxs
b.fxo
c.t1/e1
d.e&m
Ans d
and few more questions, i can't recollect
Thanks to all the other comments and all d best for guys to write this exam
1-firware image file
ans-.sbn
2.one question regarding variable dial plan
3.one scenerio based question
I think this dump is more than enough to pass...
Please answer the following question.
Q.1) ephone has 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....
Q.2) 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
Q.3) disadvantage of having Variable Length Dial Plan
Q.4)dsp value precedence of 001010, what is the consecutive DSCP value
Please Help. Thanks
If anyone can share 642-437 latest study guide.
Thanks
bandwidth required for a call (8 + 20 + 2) bytes/packet * 50 packets/second * 8bit/bytes = 12000 bps = 12kbps
So for the bandwidth of 64 kbps, we can have 5 calls continuously.
Got less marks in CUBE & Voice and Video section, please check the answer for those 2 section once again ...
Thanks Nagamin & Other's who have shared the exam experience here :)
ISDN <-> TCP <-> TCP <->q.921 <-> q.921 <-> q.931 this is the right answer