bingo, I think that PDF is not yet corrected completely!
test
August 30th, 2016
Give us an example! C’mon guys, this is a forum, lets discuss what you thing is wrong.
Bart
August 30th, 2016
post only questions that were on the exam.
Thank you
test
August 30th, 2016
Post what you thing is wrong and we will discuss it!
Bart
August 30th, 2016
except that there are wrong answers .60% is unnecessarily because it does not come in the exam,
test
August 30th, 2016
Did u try the exam, which dump you used?
Bart
August 30th, 2016
I failed twice 🙁
test
August 30th, 2016
Which dump you used? Did u review all of the comments here?
Bart
August 30th, 2016
161q dump is valid ,but many answers in the dump 161q are wrong
mario
August 31st, 2016
Hello!
Would someone share the MANICSHEEP vce?
mariocollab AT gmail DOT com
😀
sipPbx
August 31st, 2016
Bill,
An engineer is performing an international multisite deployment and wants to create an effective backup method to access TEHO destinations in case the call limit triggers. Which technology provides this ability?
A. AAR
B. SRST
C. CFUR
D. LRG
Guru’s Answer: A
Correct Answer: A (Maybe)
Explanation: I’m kind of tempted to go for LRG here, as generally you would enable TEHO with a route to the remote destination, then LRG as a backup. However the mention of “in the case the call limit triggers” makes me think they are pointing you toward AAR. I’m unsure.
I think it should be D (LRG). As I mentioned earlier AAR works only for internal numbers and uses external phone mask + AAR group to prefix the internal number to real PSTN number. Yep, you are correct that there is the confusing “call limit trigger” but you are using the WAN (as well as internal dialing) for TEHO numbers and if the connection is down or limited based on the location it will re-route the TEHO (PSTN) numbers via the LGR to the local VG.
SquierZ
August 31st, 2016
Hi Kerca,
Would you please share us 356Q passleader dump in vce file. (With VCE if possible)
I going to test on Sep, 2 2016, I will let all of u know the result.
I have studied CIPTV2 text book, and read 168Q, 355Q already.
kerca
August 31st, 2016
Hi All
Please post your email accounts to send the CVE file.
Thanks
sipPbx
August 31st, 2016
Hi Kerca, fakedummt gmail com
The other option is to upload it in Dropbox.
Sumadia
August 31st, 2016
Hi Kerca
mlinarevsan10 (at) gmail (dot) com
SquierZ
August 31st, 2016
Hi Kerca,
squierz(at)live(dot)com
Thanks in advance
sipPbx
August 31st, 2016
fakedummt (at) gmail (dot) com
mario
August 31st, 2016
Hi Kerca,
Much appreciated.
mariocollab AT gmail DOT com
Inzerat111
August 31st, 2016
Kerca,
Please share it with me as well: inzerat111 AT gmail DOT com
Thank you.
Bill
August 31st, 2016
@sipPbx – Yes I agree. I actually just passed this exam today, and got this very question. However, in the real exam, it doesn’t mention the call limit and the question is clearer, so the answer is LRG.
Inzerat111
August 31st, 2016
@Bill
What materials have you used? Could you please share it with us? Thank you.
Inzerat111
August 31st, 2016
@Bill
…one more thing…CONGRATULATIONS!
Abdulusa
August 31st, 2016
Bill which one is valid 356 or 161 ?
DP
August 31st, 2016
Kerca,
Please share it with me as well: pchinnamnaidu (AT) gmail (DOT) com
Thank you.
Bill
August 31st, 2016
I didn’t really use any of the previous ones because they all seemed out of date or wrong. The 356 PDF that Kerca posted seems to have all the correct questions and looks like the answers have been updated.
tonanog
August 31st, 2016
Hi Kerca,
Please send it to me – {email not allowed}
Thanks in advance!
tonanog
August 31st, 2016
Sorry email is tonanog_r at yahoo.com
Thanks!
Kerca
August 31st, 2016
Hi
I just share the file with the people who send me your email.
Thanks
amy
August 31st, 2016
Hi Kerca,
please send to me as well on email
katkadumpy AT gmail DOT com
thanks a lot
Elie
August 31st, 2016
Hi Kerca,
Please send it to me on my email
elie DOT gerges DOT 91 AT hotmail DOT com
Many thankss
JOSES
August 31st, 2016
Hi Kerca,
Please send it to me on my email
jsilvacos AT hotmail DOT com
Many thankss
Abdulusa
August 31st, 2016
Kerca
send your file to macieto at gmail dot com pls.
The Dark Night
August 31st, 2016
Hi there Bill, so you mean if we can study the 356q file only one will pass? Are you sure that the questions and answers are corrected . Please advise?
mario
September 1st, 2016
Hi Kerca,
My version of vcm doesnt open the vce you shared.
Thanks though
lewis
September 1st, 2016
Please also send to louispanganiban at gmail dot com Thank you
mario
September 1st, 2016
Many thanks to Kerca!
sipPbx
September 1st, 2016
Anyone downloaded the file from Kercal? I receive:
You don’t have access to this content. You’re signed in as….request access.
DP
September 1st, 2016
Many thanks to Kerca
sipPbx
September 1st, 2016
@DP, did u receive the file? I have no access via dropbox, could you please send as an attachment to fakedummt (at) gmail (dot) com
Matthews
September 1st, 2016
Hi Kerca,
Could you please share the VCE 356q with me?
mathews DOT lopes AT gmail DOT com
Thanks a lot!
micollab
September 1st, 2016
@ Kerca
Could you please also share the VCE 356q with me?
Micollab AT Yahoo DOT com
Thanks a lot!
Kerca
September 1st, 2016
Hi
I just send a new email with the link of VCE file.
Thanks
tonanog
September 1st, 2016
seems that the answers on kerca’s VCE has been updated with the correct one. However, i have question on this one:
Q80 (on 161q)
What is the correct value to use for the “DSCP for TelePresence Calls” Cisco CallManager service
parameter?
A.28 (011100)
B.34 (100010)
C.41 (101001)
D.46 (101110)
On VCE the answer is B. But if you set the DSCP value for telepresence on CUCM to 34, then the class-map that will capture the traffic will be VIDEO not the class-map that is supposed to be for TELEPRESENCE. I think the correct answer is A.
@tonanog,
The exhibit is for a router and not CUCM.
Having said that however. I agree that B is incorrect.
You are asked to state the correct value, for DSCP for telepresence, that is shown in the exhibit.
The value in the exhibit for class-map: telepresence is af32 (or DSCP 28)
The correct DSCP value for telepresence is 32 as stated above by @test
However, this value is not listed in any of the answers.
So I am thinking that since I cannot put the answer as DSCP 28; I would not have corrected the value in doing so. Then I would keep the answer as B and make a comment in the exam if time permits.
Everyone who requested the VCE has been sent.
Please come back and tell us how the exam went.
All the best.
Manic 🙂
MANICSHEEP
September 2nd, 2016
DO NOT USE 355!!!
There’s a lot of incorrect answers and you will only
be confusing yourself even further.
Focus on the 161 that i have shared and you will be okay.
USE 161 !!!
sipPbx
September 2nd, 2016
@MANICSHEEP,
Kerca shared a new one .VCE with 356Q&A and it seems that all of the questions are corrected there! You have already shared a .VCE with 70 questions but there were also wrong question .
sipPbx
September 2nd, 2016
One of the wrong questions in your .VCE version is;
@MANICSHEEP, you have a question at your dump:
Which commands are needed to configure CME in SRST mode?
– call-manager-fallback and srst mode
– telephony-service and srst mode
– telephony-service and moh
– call-manager-fallback and voice translation rule
Based on what you have selected answer D? The question says CME in SRST mode which is configured with telephony-service and srst mode?
Abdulusa
September 2nd, 2016
@MANICSHEEP
356q actually includes all questions from 161q so I think we re good.
SquierZ
September 2nd, 2016
Hi All,
As i promised you, Today just passed exam with score 959!!
I highly recommended you to use dump 356Q or 358Q provided by Kerca.
All of the question and answers are same in dump. but I think some of answers are valid about 96% so that why I got 959 not 1000.
Wish you all pass this exam too.
Sumadia
September 2nd, 2016
SquierZ Congratulation!!!
Please your opinion.
In a distributed call processing network with locations-based CAC, calls are routed to and from
intercluster trunks. Which trunk type is implemented in this network?
A.intercluster trunk with gatekeeper control
B.intercluster trunk without gatekeeper control
C.SIP trunk
D.h225 trunk
Thanks a lot!
sipPbx
September 2nd, 2016
I think its D, but most probably Cisco means A.
Abdulusa
September 2nd, 2016
“Cisco CallManager supports two models of CAC: location-based CAC for centralized call-processing environments and gatekeeper-based CAC for distributed call-processing environments. Locations-based CAC is configured on the Cisco CallManager and allocates a specific audio and video bandwidth amount to each location. IP Phones are then assigned to their respective locations on a per-device basis.
In a distributed call-control environment, multiple Cisco CallManager clusters can exist. This design requires an independent authority that can control the WAN bandwidth and call routing between clusters. The H.323 gatekeeper is used to satisfy this requirement. Gatekeepers are configured in the IOS software and added to the Cisco CallManager configuration to support gatekeeper-controlled trunks. These trunks are then included in the Cisco CallManager route plan to route outside of the cluster.” J. Cioara 😛
So I think if the question states that it is a distributed call processing, above quote confirms answer “A” – Right ?
sipPbx
September 2nd, 2016
Yep, but If we want to be correct “A” is only for CUCM version 3.1 and earlier and H.225 is for 3.1 and later :). From Cisco point of view it should be A.
Abdulusa
September 2nd, 2016
Yes, to be more correct – answer D should say ” h225 trunk gatekeeper-controlled ” to become fully correct.
tonanog
September 2nd, 2016
Hi Micollab,
i see your point. However, the exhibit shows that the router has a customized class-map for Telepresence traffic wherein instead of 32 it uses 28. If you are going to change the dscp value on CUCM to 32 or 34, then the Class-map for TELEPRESENCE will be ignored. So if the purpose of this question is to “tag the TELEPRESENCE” packets in order to match the class map on the router”. Then the answer should be A.
Nour
September 2nd, 2016
Thanks to send me the valid dump eng.nurazmy (at) gmail (dot) com
micollab
September 2nd, 2016
@tonanog,
I now agree with you. Since the exam ask you for the correct DSCP value to set on the CUCM, then for it to match the router you would have to change the DSCP vlaue to 28 on the CUCM. So that would explain why the standard DSCP value of 32 is not listed as one of the answers.
interface 1
10.1.1.2
telephony-service
ip source-address 10.1.1.1 secondary 10.1.1.2
Which option describes the effect of the configuration?
A. Implements Cisco United CME Redundancy
B. Configures standby Cisco Unified CME
C. Configures failover
D. Implements Cisco IOS redundancy
E. Creates dial peers
F Implements HSRP
If A is correct, then United is a typographical error and if so the intention would be “Implements Cisco U n i f i e d CME Redundancy”
Any thoughts?
sipPbx
September 3rd, 2016
OK, found one wrong answer at the dump Kerca shared…
“In a distrubuted call processing network with locations-based CAC, calls are routed to and from instercluster trunks. Which trunk type is implemented in this network?
a. intercluster trunk with gatekeeper
b. intercluster trunk without gatekeeper
c. SIP trunk
d. H225 trunk
It says B which is not correct. Look above comments.
voicechamp
September 3rd, 2016
but it did not mention any gatekeeper. How come b is wrong?
Abdulusa
September 3rd, 2016
@micollab
I think answer is – configures failover.
micollab
September 4th, 2016
Let’s discuss question 50 from Kerca’s 358q
After adding SRST functionality the SRST does not work. After reviewing the exhibits, which of the following reasons could be causing this failure?
A. Device Pool cannot be default.
B. The Cisco UCM is pointing to the wrong IPv4 address of the BR router.
C. The router does not support SRST.
D. The SRST enabled router is not configured correctly.
Correct Answer: D (and not A. as per 358q
Section: (none)
Explanation
Explanation/Reference:
The device pool is necessary for SRST. You have to specifically list your VG in those settings within the DP. There is no way to tell what is set in the DP, because
you only are given a name.
The CUCM is pointing to the correct interface of BR2 router. The only other interface is the sub interface going to the WAN.
All Cisco VGs support SRST and come with complimentary free 25 licenses for SRST mode. If it doesn’t have SRST abilities you won’t get past cm fall back
command.
The SRST configurations is missing many optional settings, but the one configuration that is necassary is what source port to be used. Normally the configuration
looks like – source ip address 1.1.1.1 port 2000. Port 2000 is also given in the SRST BR2 configurations image.
micollab
September 4th, 2016
Which statement is true when considering a Cisco VoIP environment for regional configuration?
A. G.711 requires 128K of bandwidth per call.
B. G.729 requires 24K of bandwidth per call.
C. The default codec does not matter if you have defined a hardware MTP in your Cisco Unified Communications Manager environment.
D. To deploy a Cisco H.323 gatekeeper, you must configure MTP resources on the gatekeeper and only use G.711 between regions.
I believe the correct answer should be C. and not B.
afik
September 4th, 2016
@MANICSHEEP
Send me the dump please.
afik5330406-gmail-com
MANICSHEEP
September 4th, 2016
@AFIK
The VCE and the VCE player has been shared with you.
All the best.
sipPbx
September 4th, 2016
@MANIACSHEEP,
could you please share it with me either – fakedummt (at) gmail (dot) com
sipPbx
September 4th, 2016
Refer to the exhibit. How many calls can be placed to Cluster B:
bandwidth inter zone default 64
bandwidth inter-zone zone Cluster B 48
Based on the dump the correct one is 3xG279? Each G.729 calls consumes 24kbps, correct?
sipPbx
September 4th, 2016
My mistake – its 16kbps. So this is the correct answer.
The Bully of books
September 4th, 2016
Passed thanks guys.
sipPbx
September 4th, 2016
@micollab, I think it’s C or D but not A.
sipPbx
September 4th, 2016
@The Bully of books, congrats! Give us more details dude! Which dump you have used?
The Bully of books
September 4th, 2016
I used the 356q file only, I didn’t correct anything I just studied the file only its valid and passed with 97x points. I finally bullied the 300-075 guys
The Bully of books
September 4th, 2016
Go and bully the 300-075; before it gets updated
The Bully of books
September 4th, 2016
This useful websites is going down soon so please make sure that you bully this exam before they take it down… Act like a bully be a bully and you will win
say
September 4th, 2016
Hello guys!
Can someone share the 356q dump?
sayroute (at) gmail (dot) com
I need to pass the 300-075 Implementing Cisco IP Telephony & Video, Part 2 v1.0 (CIPTV2)
Can someone share the 356q dump? (PDF and/or VCE files)
mapodres (at) hotmail (dot) com
Thank you !!
tonanog
September 5th, 2016
Hi guys,
those who already took the exam, can you verify how many choices were asked for this specific questions:
An engineer is configuring URI calling within the same cluster.
Which two actions must be taken to accomplish this configuration? (Choose two.)
A. Configure SIP route patterns.
B. Configure the directory URI partition and calling search space.
C. Associate the directory URIs to directory numbers.
D. Activate the URI service in Cisco Unified Serviceability.
E. Configure SIP trunk.
F. Assign directory URIs to users.
G. Configure the SIP profile.
H. Configure the URI service parameters.
There are at most 4 answers found on the selections based on the below link. But the dump only asked for 2.
Yes, I passed today with 1000/1000 and all the questions on the test came from the updated 356Q. Study that one and you’ll be good.
This link might help on H.323 end hosts registration question
http://www.cisco.com/c/en/us/support/docs/unified-communications/telepresence-video-communication-server-expressway/112520-pqa-112520-00.html
@Zana,@bingo
Can you put your exam questions here
for example
1q – A
.
.
55q – BC
.
.
132- D
.
.
Thank you
Bart, everything is in the PDF that Kerca posted
bingo, I think that PDF is not yet corrected completely!
Give us an example! C’mon guys, this is a forum, lets discuss what you thing is wrong.
post only questions that were on the exam.
Thank you
Post what you thing is wrong and we will discuss it!
except that there are wrong answers .60% is unnecessarily because it does not come in the exam,
Did u try the exam, which dump you used?
I failed twice 🙁
Which dump you used? Did u review all of the comments here?
161q dump is valid ,but many answers in the dump 161q are wrong
Hello!
Would someone share the MANICSHEEP vce?
mariocollab AT gmail DOT com
😀
Bill,
An engineer is performing an international multisite deployment and wants to create an effective backup method to access TEHO destinations in case the call limit triggers. Which technology provides this ability?
A. AAR
B. SRST
C. CFUR
D. LRG
Guru’s Answer: A
Correct Answer: A (Maybe)
Explanation: I’m kind of tempted to go for LRG here, as generally you would enable TEHO with a route to the remote destination, then LRG as a backup. However the mention of “in the case the call limit triggers” makes me think they are pointing you toward AAR. I’m unsure.
I think it should be D (LRG). As I mentioned earlier AAR works only for internal numbers and uses external phone mask + AAR group to prefix the internal number to real PSTN number. Yep, you are correct that there is the confusing “call limit trigger” but you are using the WAN (as well as internal dialing) for TEHO numbers and if the connection is down or limited based on the location it will re-route the TEHO (PSTN) numbers via the LGR to the local VG.
Hi Kerca,
Would you please share us 356Q passleader dump in vce file. (With VCE if possible)
I going to test on Sep, 2 2016, I will let all of u know the result.
I have studied CIPTV2 text book, and read 168Q, 355Q already.
Hi All
Please post your email accounts to send the CVE file.
Thanks
Hi Kerca, fakedummt gmail com
The other option is to upload it in Dropbox.
Hi Kerca
mlinarevsan10 (at) gmail (dot) com
Hi Kerca,
squierz(at)live(dot)com
Thanks in advance
fakedummt (at) gmail (dot) com
Hi Kerca,
Much appreciated.
mariocollab AT gmail DOT com
Kerca,
Please share it with me as well: inzerat111 AT gmail DOT com
Thank you.
@sipPbx – Yes I agree. I actually just passed this exam today, and got this very question. However, in the real exam, it doesn’t mention the call limit and the question is clearer, so the answer is LRG.
@Bill
What materials have you used? Could you please share it with us? Thank you.
@Bill
…one more thing…CONGRATULATIONS!
Bill which one is valid 356 or 161 ?
Kerca,
Please share it with me as well: pchinnamnaidu (AT) gmail (DOT) com
Thank you.
I didn’t really use any of the previous ones because they all seemed out of date or wrong. The 356 PDF that Kerca posted seems to have all the correct questions and looks like the answers have been updated.
Hi Kerca,
Please send it to me – {email not allowed}
Thanks in advance!
Sorry email is tonanog_r at yahoo.com
Thanks!
Hi
I just share the file with the people who send me your email.
Thanks
Hi Kerca,
please send to me as well on email
katkadumpy AT gmail DOT com
thanks a lot
Hi Kerca,
Please send it to me on my email
elie DOT gerges DOT 91 AT hotmail DOT com
Many thankss
Hi Kerca,
Please send it to me on my email
jsilvacos AT hotmail DOT com
Many thankss
Kerca
send your file to macieto at gmail dot com pls.
Hi there Bill, so you mean if we can study the 356q file only one will pass? Are you sure that the questions and answers are corrected . Please advise?
Hi Kerca,
My version of vcm doesnt open the vce you shared.
Thanks though
Please also send to louispanganiban at gmail dot com Thank you
Many thanks to Kerca!
Anyone downloaded the file from Kercal? I receive:
You don’t have access to this content. You’re signed in as….request access.
Many thanks to Kerca
@DP, did u receive the file? I have no access via dropbox, could you please send as an attachment to fakedummt (at) gmail (dot) com
Hi Kerca,
Could you please share the VCE 356q with me?
mathews DOT lopes AT gmail DOT com
Thanks a lot!
@ Kerca
Could you please also share the VCE 356q with me?
Micollab AT Yahoo DOT com
Thanks a lot!
Hi
I just send a new email with the link of VCE file.
Thanks
seems that the answers on kerca’s VCE has been updated with the correct one. However, i have question on this one:
Q80 (on 161q)
What is the correct value to use for the “DSCP for TelePresence Calls” Cisco CallManager service
parameter?
A.28 (011100)
B.34 (100010)
C.41 (101001)
D.46 (101110)
On VCE the answer is B. But if you set the DSCP value for telepresence on CUCM to 34, then the class-map that will capture the traffic will be VIDEO not the class-map that is supposed to be for TELEPRESENCE. I think the correct answer is A.
Yep, another wrong question fro Cisco.
VIDEO: ef41/34
Telepresence: Cs4/3voice af/46
Telepresence CS4/32
Video AF41/34
Voice EF/46
@tonanog,
The exhibit is for a router and not CUCM.
Having said that however. I agree that B is incorrect.
You are asked to state the correct value, for DSCP for telepresence, that is shown in the exhibit.
The value in the exhibit for class-map: telepresence is af32 (or DSCP 28)
The correct DSCP value for telepresence is 32 as stated above by @test
However, this value is not listed in any of the answers.
So I am thinking that since I cannot put the answer as DSCP 28; I would not have corrected the value in doing so. Then I would keep the answer as B and make a comment in the exam if time permits.
Any other comments on this one?
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/uc_system/design/guides/videodg/vidguide/qos.html#wp1059781
Everyone who requested the VCE has been sent.
Please come back and tell us how the exam went.
All the best.
Manic 🙂
DO NOT USE 355!!!
There’s a lot of incorrect answers and you will only
be confusing yourself even further.
Focus on the 161 that i have shared and you will be okay.
USE 161 !!!
@MANICSHEEP,
Kerca shared a new one .VCE with 356Q&A and it seems that all of the questions are corrected there! You have already shared a .VCE with 70 questions but there were also wrong question .
One of the wrong questions in your .VCE version is;
@MANICSHEEP, you have a question at your dump:
Which commands are needed to configure CME in SRST mode?
– call-manager-fallback and srst mode
– telephony-service and srst mode
– telephony-service and moh
– call-manager-fallback and voice translation rule
Based on what you have selected answer D? The question says CME in SRST mode which is configured with telephony-service and srst mode?
@MANICSHEEP
356q actually includes all questions from 161q so I think we re good.
Hi All,
As i promised you, Today just passed exam with score 959!!
I highly recommended you to use dump 356Q or 358Q provided by Kerca.
All of the question and answers are same in dump. but I think some of answers are valid about 96% so that why I got 959 not 1000.
Wish you all pass this exam too.
SquierZ Congratulation!!!
Please your opinion.
In a distributed call processing network with locations-based CAC, calls are routed to and from
intercluster trunks. Which trunk type is implemented in this network?
A.intercluster trunk with gatekeeper control
B.intercluster trunk without gatekeeper control
C.SIP trunk
D.h225 trunk
Thanks a lot!
I think its D, but most probably Cisco means A.
“Cisco CallManager supports two models of CAC: location-based CAC for centralized call-processing environments and gatekeeper-based CAC for distributed call-processing environments. Locations-based CAC is configured on the Cisco CallManager and allocates a specific audio and video bandwidth amount to each location. IP Phones are then assigned to their respective locations on a per-device basis.
In a distributed call-control environment, multiple Cisco CallManager clusters can exist. This design requires an independent authority that can control the WAN bandwidth and call routing between clusters. The H.323 gatekeeper is used to satisfy this requirement. Gatekeepers are configured in the IOS software and added to the Cisco CallManager configuration to support gatekeeper-controlled trunks. These trunks are then included in the Cisco CallManager route plan to route outside of the cluster.” J. Cioara 😛
So I think if the question states that it is a distributed call processing, above quote confirms answer “A” – Right ?
Yep, but If we want to be correct “A” is only for CUCM version 3.1 and earlier and H.225 is for 3.1 and later :). From Cisco point of view it should be A.
Yes, to be more correct – answer D should say ” h225 trunk gatekeeper-controlled ” to become fully correct.
Hi Micollab,
i see your point. However, the exhibit shows that the router has a customized class-map for Telepresence traffic wherein instead of 32 it uses 28. If you are going to change the dscp value on CUCM to 32 or 34, then the Class-map for TELEPRESENCE will be ignored. So if the purpose of this question is to “tag the TELEPRESENCE” packets in order to match the class map on the router”. Then the answer should be A.
Thanks to send me the valid dump eng.nurazmy (at) gmail (dot) com
@tonanog,
I now agree with you. Since the exam ask you for the correct DSCP value to set on the CUCM, then for it to match the router you would have to change the DSCP vlaue to 28 on the CUCM. So that would explain why the standard DSCP value of 32 is not listed as one of the answers.
Need some help with this one:
device 1:
interface 1
10.1.1.1
telephony-service
ip source-address 10.1.1.1 secondary 10.1.1.2
interface 1
10.1.1.2
telephony-service
ip source-address 10.1.1.1 secondary 10.1.1.2
Which option describes the effect of the configuration?
A. Implements Cisco United CME Redundancy
B. Configures standby Cisco Unified CME
C. Configures failover
D. Implements Cisco IOS redundancy
E. Creates dial peers
F Implements HSRP
If A is correct, then United is a typographical error and if so the intention would be “Implements Cisco U n i f i e d CME Redundancy”
Any thoughts?
OK, found one wrong answer at the dump Kerca shared…
“In a distrubuted call processing network with locations-based CAC, calls are routed to and from instercluster trunks. Which trunk type is implemented in this network?
a. intercluster trunk with gatekeeper
b. intercluster trunk without gatekeeper
c. SIP trunk
d. H225 trunk
It says B which is not correct. Look above comments.
but it did not mention any gatekeeper. How come b is wrong?
@micollab
I think answer is – configures failover.
Let’s discuss question 50 from Kerca’s 358q
After adding SRST functionality the SRST does not work. After reviewing the exhibits, which of the following reasons could be causing this failure?
A. Device Pool cannot be default.
B. The Cisco UCM is pointing to the wrong IPv4 address of the BR router.
C. The router does not support SRST.
D. The SRST enabled router is not configured correctly.
Correct Answer: D (and not A. as per 358q
Section: (none)
Explanation
Explanation/Reference:
The device pool is necessary for SRST. You have to specifically list your VG in those settings within the DP. There is no way to tell what is set in the DP, because
you only are given a name.
The CUCM is pointing to the correct interface of BR2 router. The only other interface is the sub interface going to the WAN.
All Cisco VGs support SRST and come with complimentary free 25 licenses for SRST mode. If it doesn’t have SRST abilities you won’t get past cm fall back
command.
The SRST configurations is missing many optional settings, but the one configuration that is necassary is what source port to be used. Normally the configuration
looks like – source ip address 1.1.1.1 port 2000. Port 2000 is also given in the SRST BR2 configurations image.
Which statement is true when considering a Cisco VoIP environment for regional configuration?
A. G.711 requires 128K of bandwidth per call.
B. G.729 requires 24K of bandwidth per call.
C. The default codec does not matter if you have defined a hardware MTP in your Cisco Unified Communications Manager environment.
D. To deploy a Cisco H.323 gatekeeper, you must configure MTP resources on the gatekeeper and only use G.711 between regions.
I believe the correct answer should be C. and not B.
@MANICSHEEP
Send me the dump please.
afik5330406-gmail-com
@AFIK
The VCE and the VCE player has been shared with you.
All the best.
@MANIACSHEEP,
could you please share it with me either – fakedummt (at) gmail (dot) com
Refer to the exhibit. How many calls can be placed to Cluster B:
bandwidth inter zone default 64
bandwidth inter-zone zone Cluster B 48
Based on the dump the correct one is 3xG279? Each G.729 calls consumes 24kbps, correct?
My mistake – its 16kbps. So this is the correct answer.
Passed thanks guys.
@micollab, I think it’s C or D but not A.
@The Bully of books, congrats! Give us more details dude! Which dump you have used?
I used the 356q file only, I didn’t correct anything I just studied the file only its valid and passed with 97x points. I finally bullied the 300-075 guys
Go and bully the 300-075; before it gets updated
This useful websites is going down soon so please make sure that you bully this exam before they take it down… Act like a bully be a bully and you will win
Hello guys!
Can someone share the 356q dump?
sayroute (at) gmail (dot) com
@micollab (or anyone else)
can you share the 161Q? TIA
mariocollab @ gmail . com
I meant @manicsheep
Passed 300-075 exam yesterday! 960/1000 marks!
I mainly learned questions from PassLeader 300-075 dumps (http://www.passleader.com/300-075.html). 100% valid now!
And, here is PassLeader 300-075 dumps: https://drive.google.com/open?id=0B-ob6L_QjGLpSlBUZHQxc19EWHM
Good Luck!
Hi all !
I need to pass the 300-075 Implementing Cisco IP Telephony & Video, Part 2 v1.0 (CIPTV2)
Can someone share the 356q dump? (PDF and/or VCE files)
mapodres (at) hotmail (dot) com
Thank you !!
Hi guys,
those who already took the exam, can you verify how many choices were asked for this specific questions:
An engineer is configuring URI calling within the same cluster.
Which two actions must be taken to accomplish this configuration? (Choose two.)
A. Configure SIP route patterns.
B. Configure the directory URI partition and calling search space.
C. Associate the directory URIs to directory numbers.
D. Activate the URI service in Cisco Unified Serviceability.
E. Configure SIP trunk.
F. Assign directory URIs to users.
G. Configure the SIP profile.
H. Configure the URI service parameters.
There are at most 4 answers found on the selections based on the below link. But the dump only asked for 2.
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/9_0_1/ccmsys/CUCM_BK_CD2F83FA_00_cucm-system-guide-90/CUCM_BK_CD2F83FA_00_system-guide_chapter_0101111.html#CUCM_TK_SBE2D597_00
Hello Kerca,
Can you share the 356q dumps too and to {email not allowed}?
Thank you so much!
Sorry can you share to
Can you share the 356q dumps too and to jmcollaboration(at)gmail(dot)com?
Thank you so much!
Yes please share the 356q.
mapodres (at) hotmail (dot) com
Thank you !!
Yes please share the 356q.
mapodres (at) hotmail (dot) com
Thank you !!
Failed today with a 706. ALL NEW QUESTIONS. 90% of the questions are new, 356q/161q is no longer valid. Canada
I’m failed today. Use Q356
Oh ! That a sad and bad new ! Sorry for your fail.
So, what is the good dumps now ?
Hi, I have failed the exam.Only 10-15 questions from 356 dumps, all other questions are new.
sorry to hear this guys. Do you remember any topics/new questions from the exam?
Sorry to hear guys. Do you remember any of the new questions?
Hi guys
This is so bad, i hope passleader gets updated the dumps, as soon as i have it i will share with you.
Regards.