Thanks Ken! Will report back once I finish reviewing.
Dhanu
September 1st, 2015
301Q is still valid.
Passed today with 959.
Lokesh
September 1st, 2015
hi Dhanu –
Could you please share the PDF or link to download the new 301Q questions for TVOICE (642-427).Thank you
Iceman
September 2nd, 2015
Thanks Dhanu for your info.
lordofdeah
September 2nd, 2015
passed the exam with 900+ pts. the 301q is valid
ICEMAN
September 2nd, 2015
Question 227 and 257 are similar with different answer!!! What is the correct?
DriverK
September 4th, 2015
Hello I was wondering if anyone would be able to explain Q 64
You have to refer to an exhibit that shows a debug isdn q931 trace for a MGCP gateway trying to dial an international call and failing.
Channel ID i = 0xA98381
Cause i = 0x82BE – Service Not allowed
The answer says that it is a CSS issue where CSS does not permit international calls. The explanation says “Look @ message channel ID i = 0x828381” and proceeds to analyse it that way (82 = public network near local user, 83 = no route to destination), but I do not see 0x828381 anywhere in the trace.
What am I missing? Thank you very much.
DriverK
September 4th, 2015
This is the question where HQ is connected to BR2 over an IP WAN and PSTN as a fallback.
HQ – CUCM 10.1.5.10
HQ – VG 10.1.110.1
BR2 – CME 10.3.130.1
x2020 is in HQ
x4001 is in Branch
x2020 can make a call to x4001 successfully
x4001 to x2020 receives a fast busy
This question is repeated at least twice.
In one the answer is “the h323-gateway-voip bind srcaddr 10.3.130.1 command has been omitted from BR2 config”
Second the answer is “A CSS has been omitted from the trunk configured to BR2”
So, which one is it, is it because the bind srcaddr missing, or is it “A CSS has been omitted…”
Personally, I believe it is the h323 issue, but any input would be welcome.
@ driver k
the answers don’t make a lot of sense sometimes as they are Asian dudes with not great english TBH.
the main point to take is the service is not allowed…if you see this on exam you know its permissions/css/corlist.
tvoice
September 4th, 2015
@ driver k
you’ll see that in the config. look under loopback for bind source add missing or there will be screen shot of the trunk config with css missing… so you’ll know on the day what one is correct
DriverK
September 4th, 2015
Thanks tvoice.
There is no config to refer to, and neither is there a screenshot of CUCM / trunk/anything else. It is just the diagram, and then the question, that’s it.
Hi guys,
There is a number of questions concerning t-shooting steps, including answers: define problem, gather facts, etc.
Answers seem to be intuitive but 301q dump includes strange – probably wrong answers.
1. While you troubleshoot an issue with audio quality, the symtoms stop. Which troubleshooting step is next?
A – Finalize the definition of the problem
B – Continue to gather all facts about the situation
C – Document the facts
D- Finalize the observation of resuts
Answer B
In my opinion answer should be C, but from the question we don’t know if the symtoms stopped as the result of implementing action plan or they just had stopped before any actions were performed. What do you think?
2. If after observing the results of your action plan the problem still remains at what point should you restart the troubleshooting process?
A- Create action plan
B – Consider the possibilities
C – Implement Action Plan
D – Gather facts
E – Define the problem
F – Observe results
G – Problem resolved
H – Document facts
Answer B
Personally I would restart the process at Observe results step. But the question includes “after observing the results phrase” so the next step in the process is jump to either create action plan or gather facts. Am I right? Maybe I am understanding the question in a wrong way. Native English speakers what do you think?
3. What do you do if after observing the result of your tshoot the problem still exists?
A-Implement action plan
B-Process problem resolved
C-observe results
D-create action plan
E-define the problem
F-Restart problem solving
G-gather facts
H-consider possibilities
Answer H
The answer should be definitely F! Am I right?
4. At what step do you restart the tshoot process if after observing the result of your tshoot the problem still exists?
A-Process problem resolved
B-Gather facts
C-observe results
D-Restart problem solving
E-implement action plan
F-consider possibilities
G-define the problem
H-create action plan
Answer E
Hmm, in my opnion you don’t restart the process at “implement action plan” step. “Restart process” is a step itself which should be performed after another step rather than at another step. In my opinion the answer should be “observe results”.
Native speakers what do you think?
Conclusion: if the answers are right I either don’t understand questions or don’t understandand troubleshooting process which seem to be logical and intuitive.
Sometimes Cisco expects answers which don’t make sense or dumps include wrong answers 🙂
Please share your experience and or knowledge.
Thanks
“Create an action plan. Your plan should begin with the most likely cause. The plan should only change one variable at a
time. It is crucial to change or troubleshoot only one area at a time to rule out a potential resolution.
5. Implement the action plan. Perform your action plan one action at a time. Watch carefully to see whether the desired results
have been accomplished.
6. Check the results. Check to see whether problem was resolved. Use the same tools used to find the problem.
7. Analyze the results. Has the issue been resolved? If yes, go to Step 8. If no, go back to Step 4.”
so i think create action plan is the correct answer.
is it needed to execute the commands or are the logs/messages given by cisco?
i have to admin that I don’t know all cli commands that are useful for troubleshooting. :/
thank you
Tommy
October 3rd, 2015
can some please advise which is the latest 301Q dump with the full name and where it can be downloaded from? I dont wanna loose all my hard earned money by failing. I really appreciate ur help. Already failed on the first attempt. 🙁
CoLe
October 3rd, 2015
learn to read …ken posted the mediafire link on 28th august
CZ
October 5th, 2015
passed today. 301q is valid. 60 questions total. a lot of questions are from the new 301 pdf.
What is the default directory format that you need to specify for Default TFTP MOH File Path field
on the Cisco Unified Communications Manager (CUCM) server?
there is another option: \\x.x.x.x\TFTPPATH.
ccnp voice finished 😀
zabalaza
October 6th, 2015
There should be no rush to finish the ccnp voice .there is a migration path from ccnp voice to collaboration .
And even if you complete ccnp voice you will still need to pass one exam of the collaboration track to get the collaboration certificate
For me the best way will be to take the: -citp1
-Capps
-Tvoice
Then take the ciptv2 of collabotation to get your certification with just four exams instead of 6 exams. With the bonus that to pass the ciptv2 you will have to go trough the new objective where you will learn the new material – which is for me a good bridging
CoLe
October 6th, 2015
no rush to finish? xD
i wish good luck …now you have 2 weeks to get the 3 ccnp voice exams!
by the way: 6 exams, why 6? there are 5 ccnp voice exams and 4 collab exams
You are right .it is late to plan anything now . But those who went trough the migration tool and plan well will write only 4 exams to get collaboration. Instead of 6 exams –
I insist it is 6 exams if you completed the ccnp voice.
To get collaboration you will need one extra .
CoLe
October 6th, 2015
Oh ok, wasn’t clear for me.
I think the migration is a good way for anyone who have another exam from the ccnp voice way, but who haven’t any exams, should make only the collab exams.
Maybe it was easier, but on 20th october, the exams will update and in this case, there is no difference on difficulty for “newbies”. 🙂
What are the three acceptable values for one-way delay, jitter, and packet loss in a VoIP network?
(Choose three.)
A. 0-400 ms for delay
B. 1 packet loss
C. 20 ms for jitter
D. 0-150 ms for delay
E. 1 percent packet loss
F. 30 ms for jitter
Answer: A,E,F
Congrats Anja.
will you share the material that you’ve used? the link is the website for purchase.
Ian
October 16th, 2015
Hi Nina, is 300-080 dump valid?
zabalaza
October 17th, 2015
@tommy
I am sorry . It is only today that I came back to the site – but I will say you are lazy . The link to download the exam is above and the size less than 20 Mb. . .
Come on !!!
Billy
October 20th, 2015
Passed today. 300-075 still valid. Nina Thank you so much.
Nick
October 20th, 2015
642-427 Passed today 96X marks CCNP VOICE CERTIFIED next CCNP Collaboration.
Kochański
October 24th, 2015
NOTE: 642-447 CIPT1 Last day to test: October 20, 2015
And, I found some useful 300-070/300-075/300-080/300-085 study materials:
peoples you should download latest dumps from this link
” tiny. cc/9tut “
link
November 24th, 2015
What Link?
random
November 27th, 2015
there is a dump out 118 questions, not sure if its valid or not yet 300-080
Hojane
November 30th, 2015
New 300-070 Exam Questions and Answers Updated Recently:
NEW QUESTION 72
What are the two benefits of using SIP dial rules on an IP phone? (Choose two)
A. The phone can initiate dialing without sending any signaling messages to Cisco Unified CommunicationsManager.
B. The phone can detect invalid numbers and play a reorder tone without sending any signaling messages toCisco Unified Communications Manager.
C. If dialed digits match an entry of a SIP dial rule, the dialed string is sent in a single SIP 200 OK message toCisco Unified Communications Manager.
D. If Cisco Unified Communications Manager requires more digits, KPML can be used to send the remainingdigits from the SIP phone to Cisco Unified Communications Manager one-by-one.
E. If Cisco Unified Communications Manager requires more digits, another en bloc message is used to sendthe remaining digits from the SIP.
Answer: BD
NEW QUESTION 73
Which three Cisco Unified CallManager configuration steps are required to support third party SIP phones? (Choose three.)
A. Configure the device in Cisco Unified CallManager.
B. Change the proxy address in the SIP phone to an IP address or the Fully Qualified Domain Name (FQDN)of Cisco Unified CallManager.
C. Associate the device with the end user.
D. Configure the phone with the TLS username and password.
E. Configure the end user in Cisco Unified CallManager.
F. Add the MAC address of the Cisco Unified CallManager server to the SIP phone configuration page.
Answer: ACE
NEW QUESTION 74
Which four software based media resources require that the Cisco IP voice media stream Application be activated?
A. MOH.
B. SIP.
C. H.323 Gateways.
D. Annunciator.
E. Gatekeeper.
F. MTP.
G. Audio conferencing.
Answer: ADFG
NEW QUESTION 75
Which protocol is recommended to be used between Cisco Unified Communications Manager and the voice gateway to simplify the dial plan?
A. SIP.
B. SCCP.
C. H323.
D. RSVP.
E. MGCP.
Answer: E
NEW QUESTION 76
Which protocol can Cisco Unified Communications Manager not use to monitor the status of the gateway?
A. H.323
B. SIP
C. MGCP
D. SCCP
Answer: A
NEW QUESTION 77
In a centralized call-processing solution, there are five sites connected through an IP WAN. To optimize the utilization of the IP WAN, CAC needs to be implemented. How should CAC be implemented?
A. Use a gatekeeper to control allocated bandwidth properly.
B. Use locations CAC with a single location.
C. Use a gatekeeper and Cisco Unified Border Element to provide CAC to sites that use a combination of SIPand MGCP gateways.
D. Use a gatekeeper to only control those locations that use H.323 gateways and a Cisco Unified BorderElement to control those sites that use MGCP or SIP gateways.
E. Use locations CAC with five locations in addition to Hub none.
Answer: E
NEW QUESTION 78
Which two features are advantages of deploying a cluster over WAN? (Choose two.)
A. Shared line appearances
B. CentralizedDSP resources
C. Extension mobility within the cluster
D. Scalability up to 20 sites, if thereis sufficient IP WAN bandwidth
E. Failover to co-resident Cisco UnifiedCall Manager Express platforms
Answer: AC
NEW QUESTION 79
The Ajax Corporation is designing an IP telephony network using Cisco MCS 7845 Series servers, each one capable of supporting 7500 devices. The design must meet these requirements:
– Be cost-effective
– Support up to 7500 phones
– Provide a minimal level of redundancy
Which configuration will meet Ajax Corporation needs?
A. Two Cisco Unified Communications Manager servers:
1 publisher and TFTP server combined.
1 primary subscriber.
B. Three Cisco Unified Communications Manager servers:
1 publisher and TFTP server combined.
1 primary subscriber.
1 backup subscriber.
C. Four Cisco Unified Communications Manager servers:
1 publisher.
1 TFTP server.
1 primary subscriber.
1 backup subscriber.
D. Five Cisco Unified Communications Manager servers:
1 publisher.
1 TFTP server.
1 primary subscriber.
2 backup subscribers.
Answer: B
NEW QUESTION 80
How are Cisco Unified CallManager location parameters used?
A. Assign directory numbers to devices as they connect to the IP telephony network.
B. Specify the bandwidth used for audio and video calls.
C. Implement call admission control in a centralized call processing deployment.
D. Provide alternate call routing when the primary call path is unavailable.
Answer: C
NEW QUESTION 81
Which statement regarding Cisco IP voice media streaming application is correct?
A. It should be activated on the gateway in cluster that supports the TFTP service.
B. It should be activated on the gatekeeper in cluster that supports the TFTP service.
C. It should be activated on the node in cluster that does not support the TFTP service.
D. It should be activated on the node in cluster that supports the TFTP service.
Answer: D
NEW QUESTION 82
Which option describes how you add software conference bridges to Cisco Unified Communications Manager?
A. By adding a Cisco Unified CM server to the cluster.
B. By adding a software conference bridge using Conference Bridge Configuration.
C. By installing DSP to a Cisco Unified CM server.
D. By reassigning other media resources to conference resources.
Answer: A
NEW QUESTION 83
Which of these media resources can be configured in Cisco Unified Communication manager? (Choose Three)
A. MOH Server
B. Voice Termination Point DSP
C. Transcoder
D. Auto Attendant
E. Conference Bridge
Answer: ACE
NEW QUESTION 84
What is the relationship between a Region and a Location?
A. The Region codec parameter is used between a Region and its configured Locations.
B. The Region setting for a Location sets the number of audio and video calls that Location can support.
C. The codec parameter configured in the Region is only used between Regions and Location bandwidth isonly used between Locations.
D. The Region codec parameter is combined with Location bandwidth when communicating with otherRegions.
Answer: C
NEW QUESTION 85
……
BTW, These New 300-070 Exam Questions Were Updated By PassLeader, You Can Get The Newest 300-070 Dumps In PDF And VCE From — http://www.passleader.com/300-070.html
Good Luck !!!
Hojane
November 30th, 2015
New 300-085 Exam Questions and Answers Updated Recently:
NEW QUESTION 61
Which CCC action applies only to the Cisco TelePresence MPS Series?
A. block fast update requests
B. Status Duo Video
C. move back
D. release floor
Answer: A
NEW QUESTION 62
Which protocol is prioritized by Cisco TMS for conferencing?
A. SIP
B. MGCP
C. H.323
D. SCCP
Answer: C
NEW QUESTION 63
How does Cisco TMS set up a call with an endpoint that is registered with Cisco Unified Communications Manager and that is connected to Cisco VCS?
A. Cisco TMS uses an IP dialing plan.
B. Cisco TMS uses a SIP trunk.
C. Cisco TMS removes the top-level domain from calls that go through Cisco Unified Communications Manager.
D. Cisco TMS drops the call, because it can verify that the trunk between Cisco Unified Communications Manager and Cisco VCS is down.
Answer: B
NEW QUESTION 64
On Cisco TelePresence Management Suite, which two call control devices are supported for call control and setup? (Choose two.)
A. Cisco Unified Communications Manager Express
B. Cisco TelePresence Video Communication Server
C. Cisco Unified Border Element
D. Cisco Unified Communications Manager
E. any SIP-capable call control system
Answer: BD
NEW QUESTION 65
Which three are valid system connectivity statuses for systems that are automatically added to Cisco TelePresence Management Suite? (Choose three.)
A. Inaccessible
B. Failed
C. Connected
D. Reachable on Public Internet
E. Behind Firewall
F. Online
G. Remote Site
Answer: ADE
NEW QUESTION 66
To have a Provisioning menu section available on Cisco TelePresence Management Suite, what must you do?
A. You must configure an active ISDN zone.
B. You must install Cisco TelePresence Management Suite Provisioning Extension and activate it on the system.
C. You must license Cisco TelePresence Management Suite Analytics Extension.
D. You must set up Cisco TelePresence deployment to support Cisco CMR Hybrid.
Answer: B
NEW QUESTION 67
Which two actions must you take to enable provisioning on Cisco TMS and Cisco VCS? (Choose two.)
A. Apply a Device Provisioning option key on Cisco VCS.
B. Enable provisioning on Cisco VCS under Applications > Provisioning.
C. Configure Users, FindMe, Phonebooks, and Devices individually, as these all have different settings.
D. Configure a separate server to run provisioning extension services.
E. Ensure that a service account is able to connect.
F. Set the polling intervals on Users, FindMe, Phonebooks, and Devices to the same value, in order for the synchronization to work.
Answer: AE
NEW QUESTION 68
What must you do to activate the provisioning feature on a Cisco VCS endpoint?
A. Install the Device Provisioning option key.
B. Set a SIP trunk between Cisco VCS and Cisco Unified Communications Manager.
C. Add the Cisco VCS on the Cisco TMS, and enable Cisco Extension Mobility.
D. Install Cisco VCS Starter Pack Express.
Answer: A
NEW QUESTION 69
Refer to the exhibit. Which statement about the jabber-config.xml partial output is true?
A. 10.255.10.10 is the LDAP server address.
B. The client connects to the directory server by using SSH.
C. 10.255.10.10 is the Cisco Unified Communications Manager IM and Presence Service address.
D. The client connects to the directory server by using HTTPS.
Answer: D
NEW QUESTION 70
The UDS for contact resolution is enabled in the service profile configuration of the Cisco Unified Communications Manager. Which action can you execute with your Jabber Desktop?
A. modify users
B. search users
C. change users
D. delete users
Answer: B
NEW QUESTION 71
Which profile must you add to an end user record, if the end user wants to access voicemail and wants to control a deskphone via CTI with Cisco Jabber?
A. remote destination profile
B. device profile
C. SIP profile
D. service profile
E. phone service
Answer: D
NEW QUESTION 72
……
BTW, These New 300-085 Exam Questions Were Updated By PassLeader, You Can Get The Newest 300-085 Dumps In PDF And VCE From — http://www.passleader.com/300-085.html
Good Luck !!!
Hojane
November 30th, 2015
New 300-080 Exam Questions and Answers Updated Recently:
NEW QUESTION 76
Phone A is able to dial the directory number of Phone B and complete a call. However, when Phone B dials the directory number of Phone A, Phone B receives a fast-busy tone. What is causing this issue?
A. Phone A and Phone B are in different partitions.
B. Phone B does not have Phone A in its partition.
C. Phone B is not in the CSS of Phone A.
D. Phone B does not have the CSS of Phone A in its partition.
E. Phone B is not registered.
F. Phone B has the incorrect CSS to dial Phone A.
Answer: F
NEW QUESTION 77
An inbound call from the PSTN is not reaching the directory number that it is calling. When the PSTN phone calls the correct DID, only a dial tone is heard. Which command resolves this issue?
A. (config-dial-peer)#direct-inward-dial
B. (config-controller)# no provide-outside-dialtone
C. (config-if)#no dial-tone
D. (config-dial-peer)# no dial-tone
E. (config-if)#direct-inward-dial
F. (config) allow inbound dial-peer 1
Answer: A
NEW QUESTION 78
When a caller dials 9 plus an external seven-digit number, the caller hears a fast-busy tone after a period of silence. What is causing the silence?
A. There is no dial route for 9XXXXXXX on Cisco Unified Communications Manager.
B. The gateway is not dropping the leading 9, and the PSTN fails.
C. The T302 timer is waiting to expire.
D. The caller does not have the PSTN partition in the CSS.
E. The caller dialed the wrong number.
F. To dial successfully, the caller must enter a Forced Authorization Code.
Answer: C
NEW QUESTION 79
In a single-site deployment model, the internal endpoints are unable to dial from one to the other. What are two possible causes? (Choose two.)
A. The PSTN gateway is not configured.
B. The called endpoint does not have the SIP trunk enabled.
C. The called endpoint is not registered.
D. The calling endpoint is not in the CSS of the called endpoint.
E. The called endpoint is not in the partition of the calling endpoint.
F. The calling endpoint is not configured for the correct CoS.
Answer: CF
NEW QUESTION 80
What is the default interval for SAF hello packets?
A. 15 seconds
B. 15 seconds on links with speeds that are slower than T1 speeds
C. 40 seconds
D. 40 seconds onlinks with speeds that are slower than T1 speeds
E. 60 seconds
F. 60 seconds onlinks with speeds that are slower than T1 speeds
Answer: F
NEW QUESTION 81
Which configuration can be dynamically set using the Cisco Unified Communications Manager Device Mobility feature?
A. phone model and protocol
B. SRST reference and directory number
C. CSS and local gateway
D. partition and CSS
E. media resources and permanent bridges
Answer: C
NEW QUESTION 82
Which configuration is required on Cisco TelePresence Server, in order to support 1080p resolution?
A. Screen licenses must be configured.
B. Cisco TelePresence Server must be in remotely managed mode.
C. Cisco TelePresence Server must be in HD mode.
D. Cisco TelePresence Server must be configured with Cisco TelePresence Conductor.
E. Cisco TelePresence Server must be in Full HD mode.
Answer: E
NEW QUESTION 83
A user is dialing an external PSTN number with a prefix of 01 from a Cisco TelePresence SX10 Quick Set in a Cisco VCS environment. In the past, the Cisco VCS and the ISDN gateway were correctly configured with a prefix of 01, but the calls are now failing. What are three possible causes? (Choose three.)
A. The Cisco VCS Control is down.
B. The interworking setting is turned off.
C. The audio feature in the Cisco TelePresenceSX10 is turned off.
D. The SIP trunk is not configured on the gateway.
E. 01 is not a valid prefix.
F. ISDN is not enabled on the Cisco TelePresenceSX10.
G. The Cisco TelePresenceSX10 is not registered to the Cisco VCS Control.
H. The Cisco TelePresenceSX10 is not registered to the Cisco Express C.
Answer: ABG
NEW QUESTION 84
When the command utils dbreplication status is executed on the Cisco Unified Communications Manager CLI, which step should be taken next to check the database replication status?
A. View the activelog file.
B. Run the same command on all nodes of the cluster.
C. Restart the Cisco CallManager service.
D. The command utils dbreplication runtimestate must be run on the publisher.
E. The command utils dbreplication runtimestate must be run on the subscriber.
Answer: A
NEW QUESTION 85
Which tool can you use to see SAF advertisements in Cisco Unified Communications Manager?
A. Cisco Unified Real-Time Monitoring Tool
B. show eigrp neighbors command
C. debug eigrp commands
D. Terminal Monitor
Answer: A
NEW QUESTION 86
URI dialing is enabled between two clusters with the default options. The engineer that set up the URI dialing verified that all was working properly. However, a user from one of the clusters cannot dial using URI to a user in the same cluster. What do you do to resolve this issue?
A. Verify the password that is used by the authentication under Intercluster Lookup Service configuration.
B. Find out if the URI address of the called user has a capital letter in the URI string.
C. Verify that Intercluster Lookup Service is set up correctly.
D. Verify USN Data Synchronization Status.
Answer: B
NEW QUESTION 87
……
And, These New 300-080 Exam Questions Were Updated By PassLeader, You Can Get The Newest 300-080 Dumps In PDF And VCE From — http://www.passleader.com/300-080.html
Good Luck !!!
Anon
November 30th, 2015
There does appear to be a new 118 dump for 300-080 out in the internets. Anyone have a chance to verify that it’s valid?
Anonnano
November 30th, 2015
Yes a .pdf or .vce file would be great. Do appreciate the updated dump info though Hojane.
Ace
December 4th, 2015
Hi Guys,
I’m reviewing the questions on the new 300-080 dump (118q). There’s a drag and drop item question no.116 which have wrong answers.
1. One-way audio or video A. Different system manufacturer
2. Pixelation, smearing or pulsing B. Firewall with packet inspection enabled
3. Degraded video quality C. Very high noise level
4. Codec no self-view D. Packet loss
5. Echo issues E. Main source is not main camera
On the dump the answer is:
A-2
B-4
C-1
D-5
E-3
The answer should be: (for me, and need your confirmation is this is correct)
A-3
B-1
C-5
D-2
E-4
And here’s my explanation and my opinion:
A. Different system manufacturer = 3. Degraded video quality
– For example a Cisco codec using H264 and other manufacturer using other video compression, during the call the video quality will be affected with mismatch video codecs with poor video quality. This is issue is encountered usually in a multipoint call hosted by Cisco MCU and even in Cisco endpoints.
B. Firewall with packet inspection enabled = One-way audio or video
– Firewall is the culprit for one way audio and video if media ports are not properly defined and opened on the firewall. Also caused by ALG enabled in firewall for deep inspection of SIP and H323 packets.
C. Very high noise level = 5. Echo issues
– Setting the audio gain level too high will add noise and of course very high audio levels will result to echo.
D. Packet loss = Pixelation, smearing or pulsing
– High percentage of Packet loss will affect the quality of video and audio. Pixelation in video and sometimes the audio is chocking when packet loss is experienced.
E. Main source is not main camera = 4. Codec no self-view
– When the Camera is not set as the main source, there will be no self-view so you cannot see your local video (near end).
Anon
December 4th, 2015
I agree with you Ace. That is the only issue I saw with the dump though.
Anonnano
December 4th, 2015
Thank you…
You know who I am talking about.
Sy
December 4th, 2015
Your welcome Anonnano.
iChelle
December 8th, 2015
Hi, all!
I took the 300-080 exam yesterday and passed with a good score.
Many new questions on PSTN/Cisco Unified Communications Manager, but not difficult at all.
And, the guy @Hojane has listed some new questions, all were appeared in my exam, and, I got all other new questions from the premium passleader 300-080 dumps (http://www.passleader.com/300-080.html), 100% valid!
Good Luck For All Here!
Sy
December 10th, 2015
I passed my 300-080 exam today with 9xx score. The 118q by ActulaTest dump is valid.
Goodluck guys! Preparing now for 300-085 to get my CCNP Collab cert.
Valid 118q. Passed yesterday 9xx. Let me know if this doesn’t work.
Anon
December 11th, 2015
@jack, use link to download in other post.
Anonymous
December 11th, 2015
thanks Anon for sharing but link doesn’t work
Anon
December 11th, 2015
Try and download now, sorry.
Logiurato
December 15th, 2015
examdump
serkan ozkan
December 15th, 2015
examdump
Vilius Bagdonas
December 15th, 2015
@iChelle, cool guy! Thanks for your valid comments in time! I just passed my 300-080 exam few hours ago by training PassLeader 300-080 dumps, 100% valid NOW!
Hi guys,
I passed today. 300-080 118q is still valid. Good luck
kreemo
December 15th, 2015
I passed today. 300-080 118q is still valid
Ahmad Kefaya
December 17th, 2015
i passed today for two exam 300-080 & 300-085,
300-080–>using 118 Q
300-085–>using 96 Q
wish you best all
Regards,
Rob
December 18th, 2015
A bit late to post this, but passed 300-080 earlier this week. I would Verify Questions regarding “troubleshooting registration issues” on the 118q dump. Received below a 80% on that section, so a few answeres must be off. The rest were spot on.
victoria
December 22nd, 2015
Can anyone share latest dumps 300-080 & 300-085
Thx in advance
Exam was not hard at all, for me, BUT please pay close attention to the Simulation Questions.
And, I used the premium passleader 300-080 dumps (http://www.passleader.com/300-080.html) for preparing for the exam, all new questions were available in PassLeader!
Thanks all useful and helpful comments here, and HAPPY NEW YEAR!
Thanks Ken! Will report back once I finish reviewing.
301Q is still valid.
Passed today with 959.
hi Dhanu –
Could you please share the PDF or link to download the new 301Q questions for TVOICE (642-427).Thank you
Thanks Dhanu for your info.
passed the exam with 900+ pts. the 301q is valid
Question 227 and 257 are similar with different answer!!! What is the correct?
Hello I was wondering if anyone would be able to explain Q 64
You have to refer to an exhibit that shows a debug isdn q931 trace for a MGCP gateway trying to dial an international call and failing.
Channel ID i = 0xA98381
Cause i = 0x82BE – Service Not allowed
The answer says that it is a CSS issue where CSS does not permit international calls. The explanation says “Look @ message channel ID i = 0x828381” and proceeds to analyse it that way (82 = public network near local user, 83 = no route to destination), but I do not see 0x828381 anywhere in the trace.
What am I missing? Thank you very much.
This is the question where HQ is connected to BR2 over an IP WAN and PSTN as a fallback.
HQ – CUCM 10.1.5.10
HQ – VG 10.1.110.1
BR2 – CME 10.3.130.1
x2020 is in HQ
x4001 is in Branch
x2020 can make a call to x4001 successfully
x4001 to x2020 receives a fast busy
This question is repeated at least twice.
In one the answer is “the h323-gateway-voip bind srcaddr 10.3.130.1 command has been omitted from BR2 config”
Second the answer is “A CSS has been omitted from the trunk configured to BR2”
So, which one is it, is it because the bind srcaddr missing, or is it “A CSS has been omitted…”
Personally, I believe it is the h323 issue, but any input would be welcome.
Thank you.
HQ ———– ——————————————————————————————————————–BRanch2
x2020——CUCM 10.1.5.10——VG 10.1.110.1 — {{IP WAN/PSTN}} — CME 10.3.130.1 —- x4001
@ driver k
the answers don’t make a lot of sense sometimes as they are Asian dudes with not great english TBH.
the main point to take is the service is not allowed…if you see this on exam you know its permissions/css/corlist.
@ driver k
you’ll see that in the config. look under loopback for bind source add missing or there will be screen shot of the trunk config with css missing… so you’ll know on the day what one is correct
Thanks tvoice.
There is no config to refer to, and neither is there a screenshot of CUCM / trunk/anything else. It is just the diagram, and then the question, that’s it.
301q is valid
hey .
anyone has the pdf for the 301 or 315 question ??? please inbox me on lestoncourt2@webmail.co.za
Thanks.
Hi guys,
There is a number of questions concerning t-shooting steps, including answers: define problem, gather facts, etc.
Answers seem to be intuitive but 301q dump includes strange – probably wrong answers.
1. While you troubleshoot an issue with audio quality, the symtoms stop. Which troubleshooting step is next?
A – Finalize the definition of the problem
B – Continue to gather all facts about the situation
C – Document the facts
D- Finalize the observation of resuts
Answer B
In my opinion answer should be C, but from the question we don’t know if the symtoms stopped as the result of implementing action plan or they just had stopped before any actions were performed. What do you think?
2. If after observing the results of your action plan the problem still remains at what point should you restart the troubleshooting process?
A- Create action plan
B – Consider the possibilities
C – Implement Action Plan
D – Gather facts
E – Define the problem
F – Observe results
G – Problem resolved
H – Document facts
Answer B
Personally I would restart the process at Observe results step. But the question includes “after observing the results phrase” so the next step in the process is jump to either create action plan or gather facts. Am I right? Maybe I am understanding the question in a wrong way. Native English speakers what do you think?
3. What do you do if after observing the result of your tshoot the problem still exists?
A-Implement action plan
B-Process problem resolved
C-observe results
D-create action plan
E-define the problem
F-Restart problem solving
G-gather facts
H-consider possibilities
Answer H
The answer should be definitely F! Am I right?
4. At what step do you restart the tshoot process if after observing the result of your tshoot the problem still exists?
A-Process problem resolved
B-Gather facts
C-observe results
D-Restart problem solving
E-implement action plan
F-consider possibilities
G-define the problem
H-create action plan
Answer E
Hmm, in my opnion you don’t restart the process at “implement action plan” step. “Restart process” is a step itself which should be performed after another step rather than at another step. In my opinion the answer should be “observe results”.
Native speakers what do you think?
Conclusion: if the answers are right I either don’t understand questions or don’t understandand troubleshooting process which seem to be logical and intuitive.
Sometimes Cisco expects answers which don’t make sense or dumps include wrong answers 🙂
Please share your experience and or knowledge.
Thanks
@voyteeck
will you please send me the 301 pdf at: lestoncourt2@webmail.co.za
Thanks .
Took 642-427 exam 2 days ago, scored 960/1000 points.
I used the newest 642-427 dumps, all new questions from it, if you want to try, please leave e-mail here, I will send the file to you.
Hi John Ment,
Can you please send me one copy of 642-427 dumps?
Thanks a lot,
Hendry
Email:
zyxwv_369@hotmail.com
Hi John,
Please send me the dumps to faisal.saoud@hotmail.com
hi john .
please if i can get a copy . lestoncourt2@webmail.co.za
Voyteeck
“Create an action plan. Your plan should begin with the most likely cause. The plan should only change one variable at a
time. It is crucial to change or troubleshoot only one area at a time to rule out a potential resolution.
5. Implement the action plan. Perform your action plan one action at a time. Watch carefully to see whether the desired results
have been accomplished.
6. Check the results. Check to see whether problem was resolved. Use the same tools used to find the problem.
7. Analyze the results. Has the issue been resolved? If yes, go to Step 8. If no, go back to Step 4.”
so i think create action plan is the correct answer.
Hi CZ,
Thanks for your answer !
http://www.4shared.com/file/N5EYAJGFba/642-427ExamcollectionPremiumEx.html
Anyone recently given an exam?
Part of 315q 642-427 dumps: https://drive.google.com/open?id=0B-ob6L_QjGLpVTVvb0Z2VmFrSzQ
http://wikisend.com/download/325654/642-427.pdf please let us know if you pass with this dumb.
pass today 918/1000 used 301Q.
@John Ment
Please sent dumps. Thanks!! trelee1@hmamail.com
Pass today..900+, the 301Q is valid
is it needed to execute the commands or are the logs/messages given by cisco?
i have to admin that I don’t know all cli commands that are useful for troubleshooting. :/
thank you
can some please advise which is the latest 301Q dump with the full name and where it can be downloaded from? I dont wanna loose all my hard earned money by failing. I really appreciate ur help. Already failed on the first attempt. 🙁
learn to read …ken posted the mediafire link on 28th august
passed today. 301q is valid. 60 questions total. a lot of questions are from the new 301 pdf.
What is the default directory format that you need to specify for Default TFTP MOH File Path field
on the Cisco Unified Communications Manager (CUCM) server?
there is another option: \\x.x.x.x\TFTPPATH.
ccnp voice finished 😀
There should be no rush to finish the ccnp voice .there is a migration path from ccnp voice to collaboration .
And even if you complete ccnp voice you will still need to pass one exam of the collaboration track to get the collaboration certificate
For me the best way will be to take the: -citp1
-Capps
-Tvoice
Then take the ciptv2 of collabotation to get your certification with just four exams instead of 6 exams. With the bonus that to pass the ciptv2 you will have to go trough the new objective where you will learn the new material – which is for me a good bridging
no rush to finish? xD
i wish good luck …now you have 2 weeks to get the 3 ccnp voice exams!
by the way: 6 exams, why 6? there are 5 ccnp voice exams and 4 collab exams
http://www.cisco.com/web/learning/tools/ccnp_collab/ccnp_collab_tool.html
@Cole
You are right .it is late to plan anything now . But those who went trough the migration tool and plan well will write only 4 exams to get collaboration. Instead of 6 exams –
I insist it is 6 exams if you completed the ccnp voice.
To get collaboration you will need one extra .
Oh ok, wasn’t clear for me.
I think the migration is a good way for anyone who have another exam from the ccnp voice way, but who haven’t any exams, should make only the collab exams.
Maybe it was easier, but on 20th october, the exams will update and in this case, there is no difference on difficulty for “newbies”. 🙂
Wish you all good Luck and much practice
Pass today 9XX… 301Q is valid.
passed today . 301q still walid
Link to latest CCNP Collaboration material.
https://drive.google.com/folderview?id=0B-HUe3LbqOa2X3BuUkk4YXhRaXc&usp=sharing
Thanks always welcome
What are the three acceptable values for one-way delay, jitter, and packet loss in a VoIP network?
(Choose three.)
A. 0-400 ms for delay
B. 1 packet loss
C. 20 ms for jitter
D. 0-150 ms for delay
E. 1 percent packet loss
F. 30 ms for jitter
Answer: A,E,F
Passed 642-427 exam with the valid 315q dumps — http://www.passleader.com/642-427.html (Updated: Oct 13, 2015)
Good Luck!
Congrats Anja.
will you share the material that you’ve used? the link is the website for purchase.
Hi Nina, is 300-080 dump valid?
@tommy
I am sorry . It is only today that I came back to the site – but I will say you are lazy . The link to download the exam is above and the size less than 20 Mb. . .
Come on !!!
Passed today. 300-075 still valid. Nina Thank you so much.
642-427 Passed today 96X marks CCNP VOICE CERTIFIED next CCNP Collaboration.
NOTE: 642-447 CIPT1 Last day to test: October 20, 2015
And, I found some useful 300-070/300-075/300-080/300-085 study materials:
300-070 Exam Dumps
http://www.cipt1.com/category/cisco-2/300-070-exam-dumps/
300-075 Exam Dumps
http://www.cipt1.com/category/cisco-2/300-075-exam-dumps/
P.S. These 300-075 exam questions are the newest, it seems that PassLeader just updated the 300-075 exam!
300-080 Exam Dumps
http://www.cipt2.com/category/cisco-braindump/300-080-exam-dumps/
300-085 Exam Dumps
http://www.cipt2.com/category/cisco-braindump/300-085-exam-dumps/
Hello, Could someone share the latest valid dumps?
300-070
300-075
300-080
Thanks,
osshakhatreh@hotmail.com
Is Nina’s 71q 300-080 dump valid? Any confirmation?
71q and 94q from passleader is not valid
Hi, All,
Could someone share the latest valid dump?
300-080
300-085
edisson.realpe.i@hotmail.com
thanks
Anyone pass this Exam ??
Hello,
anyone have the valid dump for 300-080 and 300-085..
please send it to me @ ahmad.m.kefaya@hotmail.com
Regards,
has anyone attempted to take 300-080?
Hello,
Anyone with valid dumps for 210-060?
Please share – Rawil-05-02@mail.ru
Thank you!
Any update from anyone on 080 and 085?
peoples you should download latest dumps from this link
” tiny. cc/9tut “
What Link?
there is a dump out 118 questions, not sure if its valid or not yet 300-080
New 300-070 Exam Questions and Answers Updated Recently:
NEW QUESTION 72
What are the two benefits of using SIP dial rules on an IP phone? (Choose two)
A. The phone can initiate dialing without sending any signaling messages to Cisco Unified CommunicationsManager.
B. The phone can detect invalid numbers and play a reorder tone without sending any signaling messages toCisco Unified Communications Manager.
C. If dialed digits match an entry of a SIP dial rule, the dialed string is sent in a single SIP 200 OK message toCisco Unified Communications Manager.
D. If Cisco Unified Communications Manager requires more digits, KPML can be used to send the remainingdigits from the SIP phone to Cisco Unified Communications Manager one-by-one.
E. If Cisco Unified Communications Manager requires more digits, another en bloc message is used to sendthe remaining digits from the SIP.
Answer: BD
NEW QUESTION 73
Which three Cisco Unified CallManager configuration steps are required to support third party SIP phones? (Choose three.)
A. Configure the device in Cisco Unified CallManager.
B. Change the proxy address in the SIP phone to an IP address or the Fully Qualified Domain Name (FQDN)of Cisco Unified CallManager.
C. Associate the device with the end user.
D. Configure the phone with the TLS username and password.
E. Configure the end user in Cisco Unified CallManager.
F. Add the MAC address of the Cisco Unified CallManager server to the SIP phone configuration page.
Answer: ACE
NEW QUESTION 74
Which four software based media resources require that the Cisco IP voice media stream Application be activated?
A. MOH.
B. SIP.
C. H.323 Gateways.
D. Annunciator.
E. Gatekeeper.
F. MTP.
G. Audio conferencing.
Answer: ADFG
NEW QUESTION 75
Which protocol is recommended to be used between Cisco Unified Communications Manager and the voice gateway to simplify the dial plan?
A. SIP.
B. SCCP.
C. H323.
D. RSVP.
E. MGCP.
Answer: E
NEW QUESTION 76
Which protocol can Cisco Unified Communications Manager not use to monitor the status of the gateway?
A. H.323
B. SIP
C. MGCP
D. SCCP
Answer: A
NEW QUESTION 77
In a centralized call-processing solution, there are five sites connected through an IP WAN. To optimize the utilization of the IP WAN, CAC needs to be implemented. How should CAC be implemented?
A. Use a gatekeeper to control allocated bandwidth properly.
B. Use locations CAC with a single location.
C. Use a gatekeeper and Cisco Unified Border Element to provide CAC to sites that use a combination of SIPand MGCP gateways.
D. Use a gatekeeper to only control those locations that use H.323 gateways and a Cisco Unified BorderElement to control those sites that use MGCP or SIP gateways.
E. Use locations CAC with five locations in addition to Hub none.
Answer: E
NEW QUESTION 78
Which two features are advantages of deploying a cluster over WAN? (Choose two.)
A. Shared line appearances
B. CentralizedDSP resources
C. Extension mobility within the cluster
D. Scalability up to 20 sites, if thereis sufficient IP WAN bandwidth
E. Failover to co-resident Cisco UnifiedCall Manager Express platforms
Answer: AC
NEW QUESTION 79
The Ajax Corporation is designing an IP telephony network using Cisco MCS 7845 Series servers, each one capable of supporting 7500 devices. The design must meet these requirements:
– Be cost-effective
– Support up to 7500 phones
– Provide a minimal level of redundancy
Which configuration will meet Ajax Corporation needs?
A. Two Cisco Unified Communications Manager servers:
1 publisher and TFTP server combined.
1 primary subscriber.
B. Three Cisco Unified Communications Manager servers:
1 publisher and TFTP server combined.
1 primary subscriber.
1 backup subscriber.
C. Four Cisco Unified Communications Manager servers:
1 publisher.
1 TFTP server.
1 primary subscriber.
1 backup subscriber.
D. Five Cisco Unified Communications Manager servers:
1 publisher.
1 TFTP server.
1 primary subscriber.
2 backup subscribers.
Answer: B
NEW QUESTION 80
How are Cisco Unified CallManager location parameters used?
A. Assign directory numbers to devices as they connect to the IP telephony network.
B. Specify the bandwidth used for audio and video calls.
C. Implement call admission control in a centralized call processing deployment.
D. Provide alternate call routing when the primary call path is unavailable.
Answer: C
NEW QUESTION 81
Which statement regarding Cisco IP voice media streaming application is correct?
A. It should be activated on the gateway in cluster that supports the TFTP service.
B. It should be activated on the gatekeeper in cluster that supports the TFTP service.
C. It should be activated on the node in cluster that does not support the TFTP service.
D. It should be activated on the node in cluster that supports the TFTP service.
Answer: D
NEW QUESTION 82
Which option describes how you add software conference bridges to Cisco Unified Communications Manager?
A. By adding a Cisco Unified CM server to the cluster.
B. By adding a software conference bridge using Conference Bridge Configuration.
C. By installing DSP to a Cisco Unified CM server.
D. By reassigning other media resources to conference resources.
Answer: A
NEW QUESTION 83
Which of these media resources can be configured in Cisco Unified Communication manager? (Choose Three)
A. MOH Server
B. Voice Termination Point DSP
C. Transcoder
D. Auto Attendant
E. Conference Bridge
Answer: ACE
NEW QUESTION 84
What is the relationship between a Region and a Location?
A. The Region codec parameter is used between a Region and its configured Locations.
B. The Region setting for a Location sets the number of audio and video calls that Location can support.
C. The codec parameter configured in the Region is only used between Regions and Location bandwidth isonly used between Locations.
D. The Region codec parameter is combined with Location bandwidth when communicating with otherRegions.
Answer: C
NEW QUESTION 85
……
BTW, These New 300-070 Exam Questions Were Updated By PassLeader, You Can Get The Newest 300-070 Dumps In PDF And VCE From — http://www.passleader.com/300-070.html
Good Luck !!!
New 300-085 Exam Questions and Answers Updated Recently:
NEW QUESTION 61
Which CCC action applies only to the Cisco TelePresence MPS Series?
A. block fast update requests
B. Status Duo Video
C. move back
D. release floor
Answer: A
NEW QUESTION 62
Which protocol is prioritized by Cisco TMS for conferencing?
A. SIP
B. MGCP
C. H.323
D. SCCP
Answer: C
NEW QUESTION 63
How does Cisco TMS set up a call with an endpoint that is registered with Cisco Unified Communications Manager and that is connected to Cisco VCS?
A. Cisco TMS uses an IP dialing plan.
B. Cisco TMS uses a SIP trunk.
C. Cisco TMS removes the top-level domain from calls that go through Cisco Unified Communications Manager.
D. Cisco TMS drops the call, because it can verify that the trunk between Cisco Unified Communications Manager and Cisco VCS is down.
Answer: B
NEW QUESTION 64
On Cisco TelePresence Management Suite, which two call control devices are supported for call control and setup? (Choose two.)
A. Cisco Unified Communications Manager Express
B. Cisco TelePresence Video Communication Server
C. Cisco Unified Border Element
D. Cisco Unified Communications Manager
E. any SIP-capable call control system
Answer: BD
NEW QUESTION 65
Which three are valid system connectivity statuses for systems that are automatically added to Cisco TelePresence Management Suite? (Choose three.)
A. Inaccessible
B. Failed
C. Connected
D. Reachable on Public Internet
E. Behind Firewall
F. Online
G. Remote Site
Answer: ADE
NEW QUESTION 66
To have a Provisioning menu section available on Cisco TelePresence Management Suite, what must you do?
A. You must configure an active ISDN zone.
B. You must install Cisco TelePresence Management Suite Provisioning Extension and activate it on the system.
C. You must license Cisco TelePresence Management Suite Analytics Extension.
D. You must set up Cisco TelePresence deployment to support Cisco CMR Hybrid.
Answer: B
NEW QUESTION 67
Which two actions must you take to enable provisioning on Cisco TMS and Cisco VCS? (Choose two.)
A. Apply a Device Provisioning option key on Cisco VCS.
B. Enable provisioning on Cisco VCS under Applications > Provisioning.
C. Configure Users, FindMe, Phonebooks, and Devices individually, as these all have different settings.
D. Configure a separate server to run provisioning extension services.
E. Ensure that a service account is able to connect.
F. Set the polling intervals on Users, FindMe, Phonebooks, and Devices to the same value, in order for the synchronization to work.
Answer: AE
NEW QUESTION 68
What must you do to activate the provisioning feature on a Cisco VCS endpoint?
A. Install the Device Provisioning option key.
B. Set a SIP trunk between Cisco VCS and Cisco Unified Communications Manager.
C. Add the Cisco VCS on the Cisco TMS, and enable Cisco Extension Mobility.
D. Install Cisco VCS Starter Pack Express.
Answer: A
NEW QUESTION 69
Refer to the exhibit. Which statement about the jabber-config.xml partial output is true?
A. 10.255.10.10 is the LDAP server address.
B. The client connects to the directory server by using SSH.
C. 10.255.10.10 is the Cisco Unified Communications Manager IM and Presence Service address.
D. The client connects to the directory server by using HTTPS.
Answer: D
NEW QUESTION 70
The UDS for contact resolution is enabled in the service profile configuration of the Cisco Unified Communications Manager. Which action can you execute with your Jabber Desktop?
A. modify users
B. search users
C. change users
D. delete users
Answer: B
NEW QUESTION 71
Which profile must you add to an end user record, if the end user wants to access voicemail and wants to control a deskphone via CTI with Cisco Jabber?
A. remote destination profile
B. device profile
C. SIP profile
D. service profile
E. phone service
Answer: D
NEW QUESTION 72
……
BTW, These New 300-085 Exam Questions Were Updated By PassLeader, You Can Get The Newest 300-085 Dumps In PDF And VCE From — http://www.passleader.com/300-085.html
Good Luck !!!
New 300-080 Exam Questions and Answers Updated Recently:
NEW QUESTION 76
Phone A is able to dial the directory number of Phone B and complete a call. However, when Phone B dials the directory number of Phone A, Phone B receives a fast-busy tone. What is causing this issue?
A. Phone A and Phone B are in different partitions.
B. Phone B does not have Phone A in its partition.
C. Phone B is not in the CSS of Phone A.
D. Phone B does not have the CSS of Phone A in its partition.
E. Phone B is not registered.
F. Phone B has the incorrect CSS to dial Phone A.
Answer: F
NEW QUESTION 77
An inbound call from the PSTN is not reaching the directory number that it is calling. When the PSTN phone calls the correct DID, only a dial tone is heard. Which command resolves this issue?
A. (config-dial-peer)#direct-inward-dial
B. (config-controller)# no provide-outside-dialtone
C. (config-if)#no dial-tone
D. (config-dial-peer)# no dial-tone
E. (config-if)#direct-inward-dial
F. (config) allow inbound dial-peer 1
Answer: A
NEW QUESTION 78
When a caller dials 9 plus an external seven-digit number, the caller hears a fast-busy tone after a period of silence. What is causing the silence?
A. There is no dial route for 9XXXXXXX on Cisco Unified Communications Manager.
B. The gateway is not dropping the leading 9, and the PSTN fails.
C. The T302 timer is waiting to expire.
D. The caller does not have the PSTN partition in the CSS.
E. The caller dialed the wrong number.
F. To dial successfully, the caller must enter a Forced Authorization Code.
Answer: C
NEW QUESTION 79
In a single-site deployment model, the internal endpoints are unable to dial from one to the other. What are two possible causes? (Choose two.)
A. The PSTN gateway is not configured.
B. The called endpoint does not have the SIP trunk enabled.
C. The called endpoint is not registered.
D. The calling endpoint is not in the CSS of the called endpoint.
E. The called endpoint is not in the partition of the calling endpoint.
F. The calling endpoint is not configured for the correct CoS.
Answer: CF
NEW QUESTION 80
What is the default interval for SAF hello packets?
A. 15 seconds
B. 15 seconds on links with speeds that are slower than T1 speeds
C. 40 seconds
D. 40 seconds onlinks with speeds that are slower than T1 speeds
E. 60 seconds
F. 60 seconds onlinks with speeds that are slower than T1 speeds
Answer: F
NEW QUESTION 81
Which configuration can be dynamically set using the Cisco Unified Communications Manager Device Mobility feature?
A. phone model and protocol
B. SRST reference and directory number
C. CSS and local gateway
D. partition and CSS
E. media resources and permanent bridges
Answer: C
NEW QUESTION 82
Which configuration is required on Cisco TelePresence Server, in order to support 1080p resolution?
A. Screen licenses must be configured.
B. Cisco TelePresence Server must be in remotely managed mode.
C. Cisco TelePresence Server must be in HD mode.
D. Cisco TelePresence Server must be configured with Cisco TelePresence Conductor.
E. Cisco TelePresence Server must be in Full HD mode.
Answer: E
NEW QUESTION 83
A user is dialing an external PSTN number with a prefix of 01 from a Cisco TelePresence SX10 Quick Set in a Cisco VCS environment. In the past, the Cisco VCS and the ISDN gateway were correctly configured with a prefix of 01, but the calls are now failing. What are three possible causes? (Choose three.)
A. The Cisco VCS Control is down.
B. The interworking setting is turned off.
C. The audio feature in the Cisco TelePresenceSX10 is turned off.
D. The SIP trunk is not configured on the gateway.
E. 01 is not a valid prefix.
F. ISDN is not enabled on the Cisco TelePresenceSX10.
G. The Cisco TelePresenceSX10 is not registered to the Cisco VCS Control.
H. The Cisco TelePresenceSX10 is not registered to the Cisco Express C.
Answer: ABG
NEW QUESTION 84
When the command utils dbreplication status is executed on the Cisco Unified Communications Manager CLI, which step should be taken next to check the database replication status?
A. View the activelog file.
B. Run the same command on all nodes of the cluster.
C. Restart the Cisco CallManager service.
D. The command utils dbreplication runtimestate must be run on the publisher.
E. The command utils dbreplication runtimestate must be run on the subscriber.
Answer: A
NEW QUESTION 85
Which tool can you use to see SAF advertisements in Cisco Unified Communications Manager?
A. Cisco Unified Real-Time Monitoring Tool
B. show eigrp neighbors command
C. debug eigrp commands
D. Terminal Monitor
Answer: A
NEW QUESTION 86
URI dialing is enabled between two clusters with the default options. The engineer that set up the URI dialing verified that all was working properly. However, a user from one of the clusters cannot dial using URI to a user in the same cluster. What do you do to resolve this issue?
A. Verify the password that is used by the authentication under Intercluster Lookup Service configuration.
B. Find out if the URI address of the called user has a capital letter in the URI string.
C. Verify that Intercluster Lookup Service is set up correctly.
D. Verify USN Data Synchronization Status.
Answer: B
NEW QUESTION 87
……
And, These New 300-080 Exam Questions Were Updated By PassLeader, You Can Get The Newest 300-080 Dumps In PDF And VCE From — http://www.passleader.com/300-080.html
Good Luck !!!
There does appear to be a new 118 dump for 300-080 out in the internets. Anyone have a chance to verify that it’s valid?
Yes a .pdf or .vce file would be great. Do appreciate the updated dump info though Hojane.
Hi Guys,
I’m reviewing the questions on the new 300-080 dump (118q). There’s a drag and drop item question no.116 which have wrong answers.
1. One-way audio or video A. Different system manufacturer
2. Pixelation, smearing or pulsing B. Firewall with packet inspection enabled
3. Degraded video quality C. Very high noise level
4. Codec no self-view D. Packet loss
5. Echo issues E. Main source is not main camera
On the dump the answer is:
A-2
B-4
C-1
D-5
E-3
The answer should be: (for me, and need your confirmation is this is correct)
A-3
B-1
C-5
D-2
E-4
And here’s my explanation and my opinion:
A. Different system manufacturer = 3. Degraded video quality
– For example a Cisco codec using H264 and other manufacturer using other video compression, during the call the video quality will be affected with mismatch video codecs with poor video quality. This is issue is encountered usually in a multipoint call hosted by Cisco MCU and even in Cisco endpoints.
B. Firewall with packet inspection enabled = One-way audio or video
– Firewall is the culprit for one way audio and video if media ports are not properly defined and opened on the firewall. Also caused by ALG enabled in firewall for deep inspection of SIP and H323 packets.
C. Very high noise level = 5. Echo issues
– Setting the audio gain level too high will add noise and of course very high audio levels will result to echo.
D. Packet loss = Pixelation, smearing or pulsing
– High percentage of Packet loss will affect the quality of video and audio. Pixelation in video and sometimes the audio is chocking when packet loss is experienced.
E. Main source is not main camera = 4. Codec no self-view
– When the Camera is not set as the main source, there will be no self-view so you cannot see your local video (near end).
I agree with you Ace. That is the only issue I saw with the dump though.
Thank you…
You know who I am talking about.
Your welcome Anonnano.
Hi, all!
I took the 300-080 exam yesterday and passed with a good score.
Many new questions on PSTN/Cisco Unified Communications Manager, but not difficult at all.
And, the guy @Hojane has listed some new questions, all were appeared in my exam, and, I got all other new questions from the premium passleader 300-080 dumps (http://www.passleader.com/300-080.html), 100% valid!
Good Luck For All Here!
I passed my 300-080 exam today with 9xx score. The 118q by ActulaTest dump is valid.
Goodluck guys! Preparing now for 300-085 to get my CCNP Collab cert.
https://filetea.me/t1s8PU2cxF3R7CSmxDATfSkmg
Valid 118q. Passed yesterday 9xx. Let me know if this doesn’t work.
@jack, use link to download in other post.
thanks Anon for sharing but link doesn’t work
Try and download now, sorry.
examdump
examdump
@iChelle, cool guy! Thanks for your valid comments in time! I just passed my 300-080 exam few hours ago by training PassLeader 300-080 dumps, 100% valid NOW!
examdump
PL’s Facebook homepage (https://tr.im/njyaW)
Good Luck!
Hi guys,
I passed today. 300-080 118q is still valid. Good luck
I passed today. 300-080 118q is still valid
i passed today for two exam 300-080 & 300-085,
300-080–>using 118 Q
300-085–>using 96 Q
wish you best all
Regards,
A bit late to post this, but passed 300-080 earlier this week. I would Verify Questions regarding “troubleshooting registration issues” on the 118q dump. Received below a 80% on that section, so a few answeres must be off. The rest were spot on.
Can anyone share latest dumps 300-080 & 300-085
Thx in advance
email: vika_pak@bk.ru
Kindly confirm whether anyone has taken the exam 300-080 & 300-085 recently
still 300-080 , 118 Q valid
@Victoria.
I have sent the dumps to your mail
I wrote the exam on 24th December and 118q was valid.
118 q are still valid, passed today
@Vilius Bagdonas
@iChelle
Can you share the valid PassLeader 300-080 dumps with me? I am going to take the exam and TIA!
Also, anyone kindly share the complete PassLeader 300-080 dumps from http://www.passleader.com/300-080.html, I only found few new questions.
Thanks again!!!
Hi,
can anyone share the valid dumps to sherio123@hotmail.com
Passed 300-080 exam yesterday!
Exam was not hard at all, for me, BUT please pay close attention to the Simulation Questions.
And, I used the premium passleader 300-080 dumps (http://www.passleader.com/300-080.html) for preparing for the exam, all new questions were available in PassLeader!
Thanks all useful and helpful comments here, and HAPPY NEW YEAR!
Hi Ibrahim,
Congrats for passing the exam.
Was it the 118q dumps that you used to pass.
Thanks
Himanshu
New 210-060 exam questions:
https://tr.im/OHWFf (Google Drive File)
New 210-065 exam questions:
https://tr.im/mg0Cy (Google Drive File)
Hi Can anyone provide pdf books on 300-070,75 CIPTV1 and CIPTV2
Implementing Cisco IP Telephony and Video, Part 1 and part 2
Hi All
please help me by sending pdf 300-080 dumps file from passleader on ( mf_ps_2014@hotmail.com )
@Ibrahim
Please kindly share that premium passleader 300-080 dumps (http://www.passleader.com/300-080.html)
My e-mail: rkqlvkittoio at outlook dot com
Thanks in advance!!!