Share your CIPT2 v8.0 Experience

Congrats! So all of your questions was from 355Q dump, correct? Could you please taka a look at the dump that Guru shared at the previous page and confirm thats the dump?

Hello, I can share 300-070, 300-075
Can you share with me 300-085?
dilt380ts (at) google (dot) com

Share please 300-075 and which version do you have – 355Q&A?

Good day Everyone as promised. I went. I fought and conquered this beast finally
with a whopping 1000/1000. Like i said earlier, i did a lot of research which cost
me a lot of money and i took a risk today and it paid off.

I will not discus what was on the exam as they might be monitoring this forum like they
did with the one they shutdown and changed the questions and making it difficult for the
rest of other test takers. If you are seriously going to write, i am willing to share my VCE player
and the 300-075 VCE and that’s all you will need to ace this exam.

How to get hold of me?

Leave your email address and i will share with you what i used and give some tips on what to look out for. Nobody deserves to lose money over these exams.

@SUMADIA

Thanks for sharing the info i asked for earlier and im sorry you failed. Do leave your mail addy and i will hook you up with the real stuff.

CCDP is next.

@MANICSHEEP
Can you send me 300-085, please?
dilt380ts (at) google (dot) com

@ANONYMOUS

Unfortunately i do not have that.

@ANONYMOUS, first of all congrats!!! Wish you take CCDP easier 🙂

Please share 300-075 at fakedummt (at) gmail (dot) com

@MANICSHEEP, first of all congrats!!! Wish you take CCDP easier 🙂

Please share 300-075 at fakedummt (at) gmail (dot) com

@MANICSHEEP

Congratulations!

mlinarevsan10 (at) gmail (dot) com

THANK YOU !!!!

@MANICSHEEP

my email is tonanog_r at yahoo.com

Thanks in advance!

Manicsheep youre the man ! macieto {at} gmail dot com pls … thank you 🙂

@MANICSHEEP

Congratulation!!

My email is {email not allowed}

@MANICSHEEP

Congratulation!!
My email is
squierz(at)live(dot)com

@MANICSHEEP
Congratulations!
sarimcdn69 (AT) gmail.com
sarimulislam (AT) hotmail.com
THANK YOU !!

Anyone received the dump from maniacsheep?

@Manicsheep, please send it to fakedummt (at) Gmail (dot) com

Hi MANICSHEEP

Could you send it please to erick_0315 (AT) hotmail.com

Regards

Hi MANICSHEEP

appreciate if you could share it to me to maximoturt (at) gmail (dot) com

Thanks

Could you please send 300-075 to flinthayden (at) gmail (dot) com

Thanks

Hi Guru Shanka

I used the 161 vce file from EC, most of its questions were there. tried competing with some answers and hence the knock.

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
Answer: A

Why not D?

Yep, and I think the correct answer is D, because AAR is used only to re-route for INTERNAL CALLS only, not for TEHO!

Hi MANICSHEEP

Congratulations on job well done, 1000/1000 on this exam is definitely a record i guess. Do you care to share your study material with me please – it is a do or die now, failed on my first attempt and i no longer trust myself

kbmooi at yahoo dot com

which 2 things do not utlise MTP
a.323 fast start
b. IPV6 -IPV4 transform
c. DTMF inband RTP-NTE (rfc2833)
d. delayed offer h.323

ONLY H.323 fast start inbound calls do not require MTP – is A is incorrect. How about BC?

Greetings to you Manicsheep and everyone.
Congratulations on your achievement.

My suggestion to you is please share your study materials via Google drive or Drop box or Mega that will help and save you time of typing numerous emails and flooding you mail box unnecessarily.

Let’s make use of technology that save time .. Cloud computing

Thanking you.

RR

The Dark Night

I have shared with everyone that sent their mail addresses.
Please do come back and tell us about your experience.
GoodLuck!

Please note that the Manager.exe and Designer.exe are crack files which you have to copy and paste on your VCE installed files to replace to original ones. Do not update the application when prompted.

To install use the visual_certexam_suite_setup.exe file.

Zana

Im not really sure, but it says on the question “when call limit triggers” which i presume refers to the bandwidth getting maxed-out. LRG is a backup mechanism for teho if the whole teho system is down. So i guess aar is the appropriate answer

Also interested!!!
sayroute (at) gmail (dot) com

@Tonanog,

Your point is valid, questions are tricky as always.

HI Manicsheep,

Sorry but i never received your email. Can you send it to me again – tonanog_r at yahoo.com?

Thanks again!

@Tonanog, @Zana

i think LRG is best answer. Within context of TEHO

“If TEHO is configured, the appropriate TEHO Gateway is used for the PSTN call. The TEHO route list can include the Default Local Route Group setting as a backup path. In this cas, if the primary (TEHO) path is not available, the gateway taht is referenced by the local route group of the applicable device pool will be used for the backup path. If the device pool selection is not static, but Cisco Unified device mobility is used, the gateway of the roaming site will be used as a backup for the TEHO path. …”

@ManicSheep

Congrats on finally get through with the exam.
Can you please also share your study material with me at

micollab at yahoo.com

Thanks very much.
Will share my experience when done

Hi Sumadia,

You maybe right. But im just confused on the question itself. You said ” if the primary (TEHO) path is not available, the gateway taht is referenced by the local route group of the applicable device pool will be used for the backup path” but on the question it doesnt say that the teho path is unavailable. It says “when the call limit triggers” which i pressume is RSVP saying the bandwidth is not enough, which is also not clear if it will activate the LRG to act as backup path.

What do you think guys?

HI Manicsheep,

the vce you sent only has 70 questions. Is this correct? Or are we missing something?

Thanks again for sending the file.

The administrator at Company X is gettinguser reports of inconsistent qualityon video calls between endpoints registered toCiscoUnifiedCommunications Manager. The administrator runs a wiretracewhile a videocall is taking place and sees that thepackets are not set to AF41 for desktop video as they should be. Whereshould the administrator look next toconfirm that the correct DSCPmarkings arebeing set?

A. onthe MGCP router at the edgeof bothnetworks B. the serviceparameters inthe VCS Control
C. the QoS service parameter in Cisco Unified Communications Manager
D.on theactual CiscophoneitselfbecausetheDSCPsettingis not partofits configurationfiledownloaded at registration
E. The setting cannot bechanged for video endpoints that areregistered toCiscoUnifiedCommunications Manager, but only whenthey areregistered to the VCS Control. ‘

Answer: C

——————-

In Cisco Unified Communications Manager, where do you configure the default bit rate for audio and video devices?

A. Enterprise Parameters
B. Region under Region Information
C. Cisco CallManager service under Service Parameter Configuration
D. Enterprise Phone Configuration

Answer: C

Correct answer is B [comment please]

C is correct.

It should be C. You can verify that when you log in to your cucm then go to service parameter

@MANICSHEEP, there are 70 questions in your version, what’s these case? This is the questions that u had on the exam or those questions are with wrong answers into the 355 dump?

@MANICSHEEP

There are 70 questions . Is your VCE file Demo or ….?

@MANICSHEEP

>I fought and conquered this beast finally with a whopping 1000/1000.

Congrats! Please share with me your study material at:

markmurk (at) mail (dot) com

@MAKMURK

CONGRATULATIONS BUDDY!! Will share the study material.

@EVERYONE

I have updated the VCE and it now has 168 questions.
All the best to you.

Hi Manicsheep and markmurk

Congragulations for the record 1000/1000 !!!!

good work done

actually i am having 355 questions brain dump and 161 questions brain dump

can you guys email me your brain dump copy so that i can analyze the differences of questions set

abhijeet14789(at)gmail(dot).com

Sorry

its abhijeet14789(at)gmail(dot)com

Thanks, C indeed correct

Hi Manicsheep!

Congaratulations for taming the beast! You’re the man!

Can you kindly share the vce to me?

Daniel.estalilla at gmail.com

@MANICSHEEP

Guru published some questions here on the forum – i’ve seen them on my last exam, is it possible to have such in vce format? Given everyone agrees with their answers, failed very bad and i can’t even comment on them

@MANICSHEEP

Just to clarify…
I only quoted you (“I fought and conquered this beast finally with a whopping 1000/1000”) and congratulated you on pasing the exam.

I haven’t taken mine yet. Please share with me your study material at:
markmurk (at) mail (dot) com

@MANICSHEEP

THANK YOU !!!!

@MANICSHEEP
Congratulations! please share the info with me as well gperez6512 -at-gmail
Thank you for your kindness to all of us.

How many Unified Mobility destination can be configured per user?

a. 1
b. 10
c. 4
d. 6

Why is the answer B not C? – see why i ask below

“Note Cisco Business Edition supports a maximum of four remote destinations per mobility user.”

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/9x/uc9x/mobilapp.html

As further shown in Figure 25-12 (Got proper answer below this figure)

“A user can have up to 10 remote destinations defined.”

Max = 10
Business Edition max = 4

Thank you, hope this helps

@Zana

10

@Zana

I will have a look at the questions and revert.

@SUMADIA

You are welcome.

@GURU

How did you get to this answers?

Which three tests can you perform to verify redundancy in the customer environment? (Choose three.)
A. Verify that all phones are registered to a second subscriber server.
B. Verify that Cisco Unified IP phones running SCCP go into SRST mode when the WAN connection is disconnected.
C. Verify that HSRP is active on the Cisco Unified Communications Manager subscriber servers.
D. Verify that media resources fail over to a secondary subscriber server when the publisher fails.
E. Verify that SCCP fallback is configured in Cisco Unified Communications Manager.
F. Verify that the H.323 redundant connection is active.
Answer: CDF

Which configuration does Cisco recommend for the peer address on the Expressway-C secure traversal zone when the Expressway-E has one NIC enabled?
A. Expressway-E internal IP address
B. Expressway-E external IP address
C. Expressway-E internal FQDN
D. Expressway-E external FQDN
Correct B

Which two options enable routers to provide basic call handling support for Cisco Unified IP Phones if they lose connection to all
Cisco Unified Communications Manager systems? (Choose two.)
A. SCCP fallback
B. MGCP fallback
C. Cisco Unified Survivable Remote Site Telephony
D. Cisco Unified Communications Manager Express
E. Cisco Unified Communications Manager Express in SRST mode
Answer: BE

A presales engineer is working on a quote for a major customer and must evaluate how many cisco VCS Expressway traversal call licenses for which to plan.
Calls to and from which three routes must the engineer include in the tally?
A. gateway
B. Cisco 9971 Endpoint
C. border controllers
D. gatekeeper
E. SIP trunk
F. VCS
Answer: D,E,F

Passed 300-075 exam yesterday!

Learned all questions from PassLeader 300-075 dumps, 100% valid now!

PassLeader 300-075 dumps: http://www.passleader.com/300-075.html

Part of that PassLeader 300-075 dumps are available here: https://drive.google.com/open?id=0B-ob6L_QjGLpSlBUZHQxc19EWHM

@Fadi Hamdan, congrats! How you have answered on the wrong questions at the dump?

@Fadi Hamdan

Congaratulations ,Can you kindly share the full vce with us.

@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?

@sipPbx I think it should be telephony-service and srst mode –
according to this article.
https://supportforums.cisco.com/document/98681/how-implement-cucme-srst-mode

Hi SipPbx and Abdulusa,

Thought i should share this – in case it helps someone:

https://supportforums.cisco.com/discussion/12009726/voice-translation-profil-under-srst

Meaning – for SRST setup to work, you need call-manager-fallback and voice-translation. This is for SRST question in the exam.

ZANA

Which two steps must you take when implementing TEHO in your environment? (Choose two.)

A.Implement local failover.
B.Implement SIP to POTS.
C.Load-balance PRI connections.
D.Load-balance route lists within the cluster.
E.Implement ICT trunks to remote locations.
F.Implement centralized failover.

@Zana
Your post is about CU SRST and the question is about CME in SRST mode, the correct answer is telephny service and srst mode

Hello Zano, yep you are right! In order to place PSTN calls you need a voice translation rules! In the questions however stands “CME in SRST mode” which is configured under the telephony-service which activates the CME into the router. Call-manager fullback is used to activate the pure SRST (w/o CME). Please find below the E-SRST (CME) configuration based on the cisco press authors:
CUCME-Router(config)# telephony-service
CUCME-Router(config-telephony)# srst mode auto-provision none
CUCME-Router(config-telephony)# srst dn line-mode dual
CUCME-Router(config-telephony)# srst ephone template 1
CUCME-Router(config-telephony)# srst dn template 1
CUCME-Router(config-telephony)# max-ephone 20
CUCME-Router(config-telephony)# max-dn 40
CUCME-Router(config-telephony)# ip source-address 10.76.108.76 port 2000
CUCME-Router(config-telephony)# moh music-on-hold.au
CUCME-Router(config-telephony)# max-conferences 4 gain -6
CUCME-Router(config-telephony)# secondary-dialtone 9
CUCME-Router(config-telephony)# system message SRST Mode

Bart its A and F probably.

@Abdulusa, i agree with A & F too.

Your view on this one please – shared mine.

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
My answer is A & D (But most correct answer is A) – your views please!
Gatekeeper-Controlled Trunks
Gatekeepers that are used in a distributed call-processing environment provide call routing and call admission control for Cisco Unified Communications Manager clusters. Intercluster trunks that are gatekeeper-controlled can communicate with all remote clusters. Similarly, an H.225 trunk can communicate with any H.323 gatekeeper-controlled endpoints including Cisco Unified Communications Manager clusters. Route patterns or route groups can route the calls to and from the gatekeeper. In a distributed call-processing environment, the gatekeeper uses the E.164 address (phone number) and determines the appropriate IP address for the destination of each call, and the local Cisco Unified Communications Manager uses that IP address to complete the call. For large distributed networks where many Cisco Unified Communications Manager clusters exist, you can avoid configuring individual intercluster trunks between each cluster by using gatekeepers.
When you configure gatekeeper-controlled trunks, Cisco Unified Communications Manager creates a virtual trunk device. The gatekeeper changes the IP address of this device dynamically to reflect the IP address of the remote device. Specify these trunks in the route patterns or route groups that route calls to and from the gatekeeper. See Cisco Unified Communications Solution Reference Network Design (SRND) for more detailed information about gatekeeper configuration, dial plan considerations when using a gatekeeper, and gatekeeper interaction with Cisco Unified Communications Manager.
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/7_1_2/ccmsys/accm-712-cm/a08trnk.html#wp1092058

Supporting doc
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/7_1_2/ccmsys/accm-712-cm/a02cac.html

@Abdulusa,

Note that Manicsheep wrote and got 1000/1000 and he selected that as an answer…”call-manager-fallback and voice translation rule”

Unless i miscouted him

@Manicsheep

Can i get your views on this two questions above please:

1. Gatekeeper question
2. SRST questions

@Zana – The question specifically mentions that calls are routed to and from intercluster trunks. To me this suggests more than one trunk i.e. a trunk from Cluster A to Cluster B and Cluster C etc. Therefore, I think A is correct, but it’s badly worded.

Also, those answers that you posted from Guru are completely wrong. HSRP in a CUCM cluster? Hmmm…

1. Correct answer is A,B,D
2. Correct answer is D. You always use the external FQDN with a single NIC then use NAT reflection for the communication.
3. C & E. MGCP fallback has nothing to do with IP Phones. It’s utilised when an MGCP gateway loses connection to CUCM and loads up a H323 stack to continue call processing.
4. This is a dubious question, because this has actually changed recently with Expressway 8.8, however, I guess the answer they are after is C, D, F.

I haven’t actually sat this exam yet, so not sure if these questions actually appear, but I would advise people not to rely on answers here. Many of the ones I’ve seen, I absolutely know to be incorrect. Have a look at the dumps as a guide for the questions, but it seems on this exam that you can’t memorise these answers and need to have a good understanding of the material (which, to me, is how it should be).

Thanks Bill,

This is my second attempt now (do or die)…what i want now is 860, will see the rest afterwards. Problem with the exam is that a correct answers matters…

@MANICSHEEP
Can you share with me as well please
jshoe102 at gmail.com
Thanks!

@Bill,

Which two options enable routers to provide basic call handling support for Cisco Unified IP Phones if they lose connection to all Cisco Unified Communications Manager systems? (Choose two.)
A. SCCP fallback
B. MGCP fallback
C. Cisco Unified Survivable Remote Site Telephony
D. Cisco Unified Communications Manager Express
E. Cisco Unified Communications Manager Express in SRST mode
Answer: BE

It says BASIC call handling support – so its should be SRST + MGCP Fallback for local PSTN connectivity. So i think the correct answers are SRST + MGCP Fallback.

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
My answer is A & D (But most correct answer is A) – your views please!

ICT with GK is used primarily in CUCM v3.1 or earlier. It should be D.

“The H.225 trunk is essentially the same as the gatekeeper-controlled ICT, except that it can work with CUCM clusters (release 3.2 and later). It also can work with other H.323 devices, such as Cisco IOS gateways (including CUCM Express), conferencing systems, and clients. This
capability is achieved through a discovery mechanism on a call-by-call basis. This type of trunk
is the recommended H.323 trunk if all CUCM clusters are at least Release 3.2.”

Another confusion question from Cisco…

Just to add to the above:

which 2 things do not utlise MTP
a. h.323 fast start
b. IPV6 -IPV4 transform
c. DTMF inband RTP-NTE (rfc2833)
d. delayed offer h.323

Guru’s Answer: A,B
Correct Answer: B,D
Explanation: Delayed offer will never use an MTP resource, whether it be H323 or SIP. H323 faststart will use an MTP resource outbound, but not inbound.

Hardware MTP requires 2 things:
a. PVDM or DSP resource
b. LTI local transcode resource
c. ref2833
d. one audio codec
e. T1 PRI card
Guru’s Answer: A,B
Correct Answer: A, D
Explanation: MTP never transcodes. While hardware MTP supports multiple codecs, it will never mix them. CUCM can use transcoders as MTPs, but it will never work the other way round.

SCCP phones register to how many nodes?
a. 1
b. 2
c. 3
d. 4
Guru’s Answer: B
Correct Answer: A
Explanaton: I don’t really like the question. Technically it “registers” to the first node in its CUCM group, but it also sends a keepalive message to the subscriber node every 60 seconds. If it loses keepalive to the primary node, then failover would occur. Therefore, my answer here would be 1, but there is a possibility Cisco could be looking for B.

VCS monitors Presence Status using what:
a. start call
b. registration
c. end call
d. call starting
Guru’s Answer: B
Correct Answer: B
Explanation: Technically, VCS will use registration, de-registration, in-call or call-ended to update presence status, so there’s a possibility that C could be correct here as well.

When you configure a globalized dial plan, in which three ways can you enable ingress gateways to process calls? (Choose three.)
A. Configure the called-party transformation settings for incoming calls on H.323 gateways.
B. Configure translation patterns in the partitions used by the gateway calling search space.
C. Configure SIP trunks between Cisco Unified Communications Manager clusters.
D. Configure a remote site device pool.
E. Configure a hunt group.
F. Configure the gateway with prefix digits to add necessary country and region codes.
Guru’s Answer: ABF
Correct Answer: ABF
Explanation: None of the others make sense. SIP tunk between clusters, device pools or hunt groups will have no impact on incoming calls.

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)
Guru’s Answer: B
Correct Answer: None
Explanation: DSCP for Telepresence Calls should be set to 32 – none of the above answers are correct. Desktop video media will use 34.

Company X has deployed a VCS Control with a local zone and a traversal client zone. To facilitate
external calls, VCS Expressway is deployed and traversal server zone is set up there. Video
endpoints inside Company X have registered but are unable to receive calls from outside
endpoints. Which option could be the cause of this issue?
A. The access control list on the VCS Control must be updated with the IP for the external users.
B. When a traversal zone is set up on VCS Control only outbound calls are possible.
C. The local zone on the VCS Control does not have a search rule configured.
D. The traversal zone on the VCS Control does not have a search rule configured.
Guru’s Answer: C
Correct Answer: Unsure
Explanation: We don’t really have enough information to tell. Perhaps in the exam there’s an exhibit or something to refer to.

Which solution is needed to enable presence and extension mobility to branch office phones during a WAN failure?
A. SRST with MGCP fallback
B. Cisco Unified Communications Manager Express in SRST mode
C. SRST without MGCP fallback
D. SRST with VoIP dial peers to Cisco Unified Communications Manager Express
Guru’s Answer: B
Correct Answer: B

Which three tests can you perform to verify redundancy in the customer environment? (Choose three.)
A. Verify that all phones are registered to a second subscriber server.
B. Verify that Cisco Unified IP phones running SCCP go into SRST mode when the WAN connection is disconnected.
C. Verify that HSRP is active on the Cisco Unified Communications Manager subscriber servers.
D. Verify that media resources fail over to a secondary subscriber server when the publisher fails.
E. Verify that SCCP fallback is configured in Cisco Unified Communications Manager.
F. Verify that the H.323 redundant connection is active.
Guru’s Answer: CDF
Correct Answer: A, B, D
Explanation: CUCM does not run HSRP, so C is absoutely wrong. SCCP fallback is configured on a gateway, not CUCM. F makes no sense. Pay attention to option A though, this makes me think the answer to the question above about the number of registrations may actually be 2, but as above, it technically doesn’t register, just sends it keepalives to make sure it’s still available.

Which three steps configure Cisco Unified Survivable Remote Site Telephony for SIP phones? (Choose three.)
A. Configure voice register pool.
B. Configure telephony service.
C. Configure a phone NTP reference.
D. Configure the SIP registrar.
E. Configure an SRST reference.
F. Configure voice register global dn.
Guru’s Answer: AEF
Correct Answer: ADF
Explanation: SIP registrar is required. Also, the command “voice register global dn” does not exist.

What is the default DSCP/PHB for TelePresence video conferencing packets in Cisco Unified Communications Manager?
A. CS4/32
B. CS6/48
C. EF/46
D. AF41/34
E. CS3/24
Guru’s Answer: A
Correct Answer: A

Which statement about the function of the “+” symbol in the E.164 format is true?
A. The “+” symbol matches the preceding element one or more times.
B. The “+” symbol matches the preceding element zero or one time.
C. The “+” symbol represents the international country code.
D. The “+” symbol represents the international call prefix.
Guru’s Answer: D
Correct Answer: D

A new DX650 IP phone with MAC address D0C7.8914.132D, IP address is 172.18.32.119 has been added to the Cisco Unified Communications
Manager, but is not registering properly. What is causing this failure?
A. The location Hub_None has not been activated.
B. Device Pool cannot be default.
C. The DX650 Phones does not support SIP.
D. The DX650’s MAC address is incorrect in the Cisco UCM.
E. The DX650 is the incorrect calling search space.
Guru’s Answer: D
Correct Answer: D
Explanation: I guess there will be an exhibit to refer to in the exam, but none of the others make sense. It has to be D.

What two tasks must be completed in order to support calls between the VCS controlled endpoints and the Cisco Unified CM endpoints?
(Choose two.)
A. Configure a SIP trunk on the Cisco Unified CM to point to the Cisco VCS.
B. Configure a neighbor zone on the Cisco VCS to point to the Cisco Unified CM.
C. Configure a neighbor zone on the Cisco Unified CM to point to the Cisco VCS.
D. Configure a SIP trunk on the Cisco VCS to point to the Cisco Unified CM.
E. Media Resource Group List.
Guru’s Answer: AB
Correct Answer: AB
Explanation: No SIP trunks are configured on VCS. The SIP trunk is configured on CUCM and then a neigbour zone on VCS, so A & B are correct.

Which two options enable routers to provide basic call handling support for Cisco Unified IP Phones if they lose connection to all
Cisco Unified Communications Manager systems? (Choose two.)
A. SCCP fallback
B. MGCP fallback
C. Cisco Unified Survivable Remote Site Telephony
D. Cisco Unified Communications Manager Express
E. Cisco Unified Communications Manager Express in SRST mode
Guru’s Answer: BE
Correct Answer: CE
Explanation: MGCP fallback is configured on a gateway to allow it to continue to process calls using H323, if it loses connection to CUCM. In this case, the phones have lost connection, so need an SRST router to register to.

Which two types of devices are affected when an engineer changes the DSCP for Video Calls
service parameter? (Choose two.)
A.DX-650
B.Cisco Jabber Desktop
C.CP-7965
D.EX-60
E.MX-200
Guru’s Answer: A,C
Correct Answer: A,B
Explanation: DX and Jabber are video endpoints. EX and MX are Telepresence endpoints, so would not be affected. C doesn’t support video therefore it can’t be correct.

A presales engineer is working on a quote for a major customer and must evaluate how many cisco VCS Expressway traversal call licenses for which to plan.
Calls to and from which three routes must the engineer include in the tally?
A. gateway
B. Cisco 9971 Endpoint
C. border controllers
D. gatekeeper
E. SIP trunk
F. VCS
Guru’s Answer: D,E,F
Correct Answer: CDF
Explanation: For a VCS Expressway, calls to or from a traversal client. Traversal clients include other VCSs, gatekeepers, Border Controllers, or traversal-enabled endpoints.

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.

Which two actions ensure that the call load from Cisco TelePresence Video Communication Server to a Cisco Unified Communications Manager cluster is shared across Unified CM nodes? (Choose two.)
A. Create a neighbor zone in VCS with the Unified CM nodes listed as location peer addresses.
B. Create a single traversal client zone in VCS with the Unified CM nodes listed as location peer addresses.
C. Create one neighbor zone in VCS for each Unified CM node.
D. Create a VCS DNS zone and configure one DNS SRV record per Unified CM node.
E. In VCS set Unified Communications mode to Mobile and remote access and configure each Unified CM node.
Guru’s Answer: AE
Correct Answer: AD
Explanation: VCS -> CUCM communcation will never utilise MRA so E is incorrect. Using a DNS zone or neighbour zone with all clients is what would achieve load balancing.

Which two statements regarding IPv4 Static NAT address 209.165.200.230 has been configured on a VCS Expressway are true? (Choose two.)
A. The Advanced Networking or Dual Network Interfaces option key has been installed.
B. VCS rewrites the Layer 3 source address of outbound SIP and H.323 packets to 209.165.200.230.
C. VCS applies 209.165.200.230 to outbound SIP and H.323 payload messages.
D. With static NAT enabled on the LAN2 interface, VCS applies 209.165.200.230 to outbound H.323 and SIP payload traffic exiting the LAN1 interface.
Guru’s Answer: AC
Correct Answer: AC

Which configuration does Cisco recommend for the peer address on the Expressway-C secure traversal zone when the Expressway-E has one NIC enabled?
A. Expressway-E internal IP address
B. Expressway-E external IP address
C. Expressway-E internal FQDN
D. Expressway-E external FQDN
Guru’s Answer: B
Correct Answer: D
Explanation: You must enter the FQDN of the Expressway-E, as it seen from outside the network, as the peer address on the Expressway-C. The reason for this is that in static NAT mode, the Expressway-E requests that incoming signalling and media should be sent to its external FQDN. This also means that the externa firewall must allow traffic from the Expressway-C to the Expressway-E’s external FQDN. This is know as NAT reflection.

Which function can be implemented without MTP resources?
A.DTMF relay conversion
B.terminating a media stream that uses the same codec
C.music on hold
D.SIP early offer
Guru’s Answer: B or C
Correct Answer: B
Explanation: Technically C can be correct in some scenarios, but there are other scenarios when MTP is used for MOH, so I’d say the “most” correct answer is B

As I said earlier, all the resources to answer these questions are freely available in configuration guides of VCS, CUCM etc. Some of the questions are badly worded though

@sippbx

“It says BASIC call handling support – so its should be SRST + MGCP Fallback for local PSTN connectivity. So i think the correct answers are SRST + MGCP Fallback.”

It specifically mentions when phones cannot reach the CUCM. MGCP fallback would only apply if an MGCP gateway lost its keepalive to CUCM. In this case C and E MUST be correct.

“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
My answer is A & D (But most correct answer is A) – your views please!

ICT with GK is used primarily in CUCM v3.1 or earlier. It should be D.”

Yes, this is a bad question. My two cents are because it refers to intercluster trunks (plural) that it must mean they are not gateway controlled trunks. And because it speciically mentions ICT, I don’t think any of the others can be correct. A SIP and H225 can offer the same features as an ICT, but technically, they’re just trunks, not ICTs. I hate the question though

However, the questions are too confused, out you should go to the head of whoever wrote them. One last:

Which three steps configure Cisco Unified Survivable Remote Site Telephony for SIP phones? (Choose three.)
A. Configure voice register pool.
B. Configure telephony service.
C. Configure a phone NTP reference.
D. Configure the SIP registrar.
E. Configure an SRST reference.
F. Configure voice register global dn.
Guru’s Answer: AEF
Correct Answer: ADF
Explanation: SIP registrar is required. Also, the command “voice register global dn” does not exist.

“voice register global dn” does not exist but u select this answer? I think ADE is the correct.

BPS-CUBE1(config)#voice register ?
dn Define dn tag
global Define global commands
pool Define pool tag

BPS-CUBE1(config)#voice register glo
BPS-CUBE1(config)#voice register global ?

BPS-CUBE1(config)#voice register global ?

BPS-CUBE1(config)#voice register global

Sorry, that was a typo 🙂 Absolutely ADE is the correct choice

And for the first question on this page – what is the correct answer from your point of view – A or B?

My answer is A, but we really don’t have enough information. Technically it could be either A or B. Mybe there is an exhibit or something to refer to in the exam.

Thanks for your time. I continue reading about SAF/CCD :). Good luck all.

Does anyone has the 355Q dump into .vce format?

sipPbx read about converting pdf to vce its really easy. You need exam formatter and vce designer which manicsheep shared the other day.

Yeah, I have tried with the 355Q dump shared here but unsuccessful :(.

I can only convert only 161Q.

Really sorry for the spam. I will highly appreciated if anyone can convert this 355Q into the VCE format. When I try to import it it doesnt pool the questions into the Exam Formater.

The network administrator of Enterprise X receives reports that at peak hours, some calls between
remote offices are not passing through. Investigation shows no connectivity problems. The
network administrator wants to estimate the volume of calls being affected by this issue.
Which two RTMT counters can give more information on this? (Choose two.)

A.CallsRingNoAnswer
B.OutOfResources
C.LocationOutOfResources
D.RequestsThrottled
E.CallsAttempted

Check this: http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/service/7_1_2/rtmt/RTMT/rtpmcm.html

CallsRingNoAnswer :This counter represents the total number of calls through a hunt list that rang but that called parties did not answer.

OutOfResources: This counter represents the total number of times that a call on a particular Cisco Unified Communications Manager through the location failed due to lack of bandwidth.

LocationOutOfResources: This counter represents the total number of times that a call through Locations failed due to the lack of bandwidth.

Wrote this morning and passed…Thanks to Manic sheep, Sumadia and everyone who partake in this forum.

To those about to write…contact the men above for reviewed answers. Read up on H.323 end host registration on VCS (new question/ might have missed it).

Scored 940

Congrats Zana, where all the question came – 355Q dump? Which answers you have selected on the wrong questions into the dump – those from Manicsheep?

@MANICSHEEP

Hello, could you please the latest materials (pdf/vce) with me as well?
Thank you so much.

inzerat111 at gmail.com

@MANICSHEEP

Send me the dump please.

kryptcruz (at) yahoo.com

@sipPbx,

Yes – used those ones. It was a do or die for me (second attempt).

TIP – review each question, seek clarity here need be (like you guys helped me). This beast can be beaten using the 161Q (haven’t seen the 355Q – but i feel that’s too much). Focus a lot on VCS, below are my results so you know which questions to give more attention:

VCS control – 70%
VCS expressway – 84%
Central call processing – 90%
Call Adminission Control 44%
Video Mobility – 86%

@MANICSHEEP
Can you please remember to send me the study material?
Micollab at Yahoo.com

Thanks much

Hello, where can we find the 161 dump? Is it available anywhere?

Hi All

I just buy the new version of passleader 300-075 with 356q updated Aug/21/16.

Passleader said to me that this file is valid now and was corrected, i put the link below to verify it together. Feel free to use it, please share your experience if this file was useful and valid.

https://www.dropbox.com/s/t1s04qq5hiosdij/300-075_356q_PassLeader.pdf?dl=0

I´m planning to do my test on sep/19/16 or before because i´m reading the guide.

Thanks

THANKS! COULD SOMEONE CINVERT IT TI VCE

I think its corrected! I checked only the wrong questions that we have already discussed here!