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QUESTION 56
Which statement is correct about AAR?
A. The end users sees, “Network Congestion Rerouting?” but AAR is otherwise transparent to the end user and works without user intervention.
B. AAR will display “not enough bandwidth” on the IP phone while it reroutes the call.
C. AAR allows calls to be rerouted because of insufficient Cisco Unified Border Element controlled bandwidth to an ITSP.
D. AAR allows calls to be rerouted due to insufficient gatekeeper controlled IP WAN bandwidth.

Correct Answer: A Section: AAR Explanation
Explanation/Reference:
Ok, CIPT2 3-76
QUESTION 57
The relationship between a Region and a Location is that the Region codec parameter is used between a Region and its configured Locations.
A. TRUE
B. FALSE

Correct Answer: B Section: Configuration Explanation
Explanation/Reference:
Ok.
QUESTION 58
Refer to the exhibit. A user in RTP calls a phone in San Jose during congestion with Call Forward No Bandwidth (CFNB) configured to reach cell phone 4085550150. The user in RTP sees the message “Not Enough Bandwidth” on their phone and hears a fast busy tone. Which two conditions can correct this issue? (Choose two.)

A. The calling phone (RTP) needs to have AAR Group value of AAR under the AAR Settings.
B. The called phone (San Jose) needs to have AAR Group value of AAR under the AAR Settings.
C. The calling phone (RTP) needs to have the AAR destination mask of 914085550150 configured under the AAR Settings.
D. The calling phone (RTP) needs to have the AAR destination mask of 4085550150 configured under the AAR Settings.
E. The called phone (San Jose) needs to have the AAR destination mask of 914085550150 configured under the AAR Settings.
F. The called phone (San Jose) needs to have the AAR destination mask of 4085550150 configured under the AAR Settings.

Correct Answer: BF Section: AAR Explanation
Explanation/Reference:
Ok
QUESTION 59
Which two statements describe RSVP-enabled locations-based CAC? (Choose two.)
A. RSVP can be enabled selectively between pairs of locations.
B. Using RSVP for CAC simply allows admitting or denying calls based on a logical configuration that is ignoring the physical topology.
C. RSVP is topology aware, but only works with full mesh networks.
D. An RSVP agent is a Media Termination Point that the call has to flow through.
E. RSVP and RTP are used between the two endpoints.

Correct Answer: AD Section: Configuration Explanation
Explanation/Reference:
Ok. See 3-51 CIPT2
QUESTION 60
You are the Cisco Unified Communications Manager in Certpaper.com. You use a remote site MGCP gateway to provide redundancy when connectivity to the central Cisco Unified Communications Manager cluster is lost. How to enable IP phones to establish calls to the PSTN when they have registered with the gateway? (Choose three.)
A. POTS dial peers must be added to the gateway to route calls from the IP phones to the PSTN.
B. The default service must be enabled globally.
C. The command ccm-manager mgcp-fallback must be configured.
D. COR needs to be configured to disallow outbound calls.

Correct Answer: ABC Section: Configuration Explanation
Explanation/Reference:
Ok, check 2-37 and 2-47
QUESTION 61
Which two configurations provide the best SIP trunk redundancy with Cisco Unified Communications Manager? (Choose two.)
A. Configure all SIP trunks with DNS SRV.
B. Configure all SIP trunks with Cisco Unified Border Element.
C. Configure all SIP trunks to point to a SIP gateway.
D. Configure SIP trunks to be members of route groups and route lists.
E. Configure all SIP trunks to allow TCP ports 5060.
F. Configure all SIP trunks to point to a gatekeeper through SIP to H.323 gateway.

Correct Answer: AD Section: Sip Trunk Explanation
Explanation/Reference:
Ok, Configure all SIP trunks with DNS SRV.
QUESTION 62
When an external call is placed from Ajax, they would like the ANI that is sent to the PSTN to be the main number, not the extension. For domestic calls, they would like 10 digits sent; for international calls, they would like to send the country code 1 and the 10 digits. How can this be accomplished?
A. Add a translation pattern to the dial peers in the gateway that adds the appropriate digits to the outgoing ANI.
B. In the external call route patterns, set the external phone number mask to the main number. Use 10 digits in the domestic route pattern and 1 followed by the main number digits in the international route patterns.
C. Use a calling party transform mask for each route group in the corresponding route list configuration.
Set the explicit 10-digit main number for domestic calls and 1 followed by the main number for the
international route patterns.
D. In the directory number configurations, set the prefix digits field to the country code and the 10 digits of the main number. This will be truncated to the 10-digit number for domestic calls and sent out in its entirety for international calls.

Correct Answer: C Section: Configuration Explanation
Explanation/Reference:
Ok, only using a calling party transformation mask can be possible.
QUESTION 63
The relationship between a Region and a Location is that the Region codec parameter is combined with Location bandwidth when communicating with other Regions.
A. FALSE
B. TRUE

Correct Answer: A Section: Configuration Explanation
Explanation/Reference:
Ok
QUESTION 64
Video calls using 384 kbps need to be supported across a gatekeeper-controlled trunk. What value should be entered into the gatekeeper to support this bandwidth?
A. 768 kbps
B. 384 kbps
C. 512 kbps
D. 192 kbps

Correct Answer: A Section: Configuration Explanation
Explanation/Reference:
Ok see 3-97 CIPT2
QUESTION 65
You are the Cisco Unified Communications Manager in Certpaper.com. After you add the Tcl paramspace command, the application can____.
A. set aside memory for application variables
B. access the data on an internal server
C. access the data on an external server
D. share parameters between different call applications

Correct Answer: D Section: Configuration Explanation
Explanation/Reference:
Ok, it could be an old question.
QUESTION 66
The following exhibit shows configs for H.323 gateway. You have been asked to implement TEHO from a remote branch office with area code 301 to the HQ office with area code 201 using Cisco Unified Communications Manager. The remote office has an MGCP gateway and the HQ office has an H.323 gateway. Once the call arrives at the HQ, it should break out to the PSTN as a seven-digit local call. Which statement about the route pattern is true?

A. route pattern should be 91201.[2-9]XXXXXX with Discard Digit as Predot and Prefix 9
B. route pattern should be 91201.[2-9]XXXXXX with Discard Digit as Predot
C. route pattern should be 91201.[2-9]XXXXXX
D. route pattern should be 9.1201[2-9]XXXXXX with Discard Digit as Predot
E. route pattern should be 9.1201[2-9]XXXXXX with Discard Digit as Predot and Prefix 9

Correct Answer: A Section: TEHO Explanation
Explanation/Reference:
Ok.
QUESTION 67
What is the default value for the Drop Ad Hoc Conference service parameter?
A. Never
B. When No On-Net Parties Remain in the Conference
C. When No Off-Net Parties Remain in the Conference
D. Drop Ad Hoc Conference When Creator Leaves

Correct Answer: A Section: Dial Plan Explanation
Explanation/Reference:
Ok System – Service Parameters – Server x.x.x.x – Service – Cisco CallManager (Active) – Clusterwide Parameters (Feature – Conference) – Drop Ad Hoc Conference – Never (default)
QUESTION 68
What is the minimum configuration required for registering a SAF client with a forwarder? (There were a couple of other choices that I do not remember)
A. SIP profile, SIP trunk, forwarder, advertising service
B. SIP profile, SIP trunk, forwarder, requesting service
C. SIP profile, SIP trunk, forwarder
D. SIP profile, SIP trunk, forwarder, advertising service & requesting service
Correct Answer: C Section: (none) Explanation

Explanation/Reference:
QUESTION 69
When configuring SIP preconditions, which of the following are true? (There were a couple of other choices that I do not remember)
A. IP phones and SIP trunks both require an RSVP agent
B. RSVP agents are only required for IP phones
C. RSVP agents are only required for SIP trunks
D. RSVP agents are only required for SIP trunks when local fallback is configured

Correct Answer: A Section: (none) Explanation
Explanation/Reference:
not sure
“The RSVP agent that is associated with the IP Phone is used for the call leg to the far-end SIP device. If QoS fallback is not enabled, the SIP trunk will never allocate an RSVP agent. If QoS fallback mode is enabled, two local RSVP agents are required in a fall-back scenario: one for the IP Phone and one for the SIP trunk. Therefore, the MRGL at the SIP trunk is only required for QoS fallback mode or for when the SIP trunk is not configured for SIP Preconditions at all but is configured to use local QoS.”
It sounds like the SIP trunk will only require an RSVP agent if local fallback is configured. That would make D the correct answer. Since I have recalled all these questions and answers from memory, it is quite possible that for this particular question, I may have split one option into two, so for instance, A & D may have been a single statement. Just try to study this part in depth so there’s no confusion during the exam in case the answers are worded differently.
QUESTION 70
How do you add a Cisco 38XX ISR router as a H.323 gateway? (There were a couple of other choices that I do not remember)
A. Select the 3800 Router series then select the exact model 38XX
B. As a H.323 gateway
C. Select the exact model 38XX

Correct Answer: B Section: (none) Explanation
Explanation/Reference:
QUESTION 71
There were two exhibits, one showing the locations configuration window with 96Kbps of bandwidth configured for calls and a second exhibit showing the RSVP configuration on the gateway. The bandwidth command under the MTP resource configuration had a question mark next to it and the question was to calculate the required bandwidth for 3 g729 calls
A. 32
B. 88
C. 72
D. 69
Correct Answer: B Section: (none) Explanation

Explanation/Reference:
There were several choices, but the correct one was 88 Kbps since 1 g729 call uses 24 Kbps, therefore 3 calls would required: 72 Kbps + 16 Kbps for RSVP
QUESTION 72
How do you configure a H.323 connection for SAF
A. the correct answer was the one with the configure ICT (non-gatekeeper) and check the box that reads: “Enable SAF”

Correct Answer: Section: (none) Explanation
Explanation/Reference:
There were several choices which I do not quite remember

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