Q1. While configuring Call Survivability in Cisco Unified Communications Manager, what step is mandatory to reach remote sites while in SRST mode?
A. Enable Cisco Remote Site Reachability.
B. Configure CFUR.
C. Enable the SRST checkbox in the MGCP gateway.
D. Configure the H.323 gateway for SRST in Cisco Unified Communications Manager.
E. Enable the Failover Service parameter.
Answer: B
Explanation:
Incorrect Answer: A, C, D, E Call Forward Unregistered (CFUR) functionality provides the automated rerouting of calls through the PSTN when an endpoint is considered unregistered due to a remote WAN link failure Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/models.html
Q2. Which two statements about remote survivability are true? (Choose two.)
A. SRST supports more Cisco IP Phones than Cisco Unified Communications Manager Express in SRST mode.
B. Cisco Unified Communications Manager Express in SRST mode supports more Cisco IP Phones than SRST.
C. MGCP fallback is required for ISDN call preservation.
D. MGCP fallback functions with SRST.
Answer: A,D
Q3. Refer to the exhibit.
The HQ site uses area code 650. The BR1 site uses area code 408. The long distance national code for PSTN dialing is 1. To make a long distance national call, an HQ or BR1 user dials access code 9, followed by 1, and then the 10-digit number.
Both sites use MGCP gateways. AAR must use globalized call routing using a single route pattern. Assume that all outgoing PSTN numbers are localized at the egress gateway as shown in the exhibit.
How many route lists and route groups should be configured for AAR at a minimum?
A. a single route list with a local route group for each site
B. two route lists and two route groups for each site
C. a single route list and four route groups for each site
D. None. The AAR CSS can point directly to the route pattern.
Answer: A
Q4. When an incoming PSTN call arrives at an H.323 gateway, how does the called number get normalized to an internal directory number in Cisco Unified Communications Manager?
A. Normalization is done by configuring the significant digits for inbound calls on the H.323 gateway configuration in Cisco Unified Communications Manager.
B. Normalization is done using route patterns.
C. Normalization is done using the gateway incoming calling party prefixes based on number type.
D. Normalization is achieved by local route group that is assigned to the H.323 gateway.
Answer: A
Q5. Which two are gatekeeper-controlled trunk options that support gatekeeper call administration control? (Choose two.)
A. H.323
B. H.245
C. H.225
D. intercluster
E. intracluster
Answer: C,D
Q6. Which two options should be selected in the SIP trunk security profile that affect the SIP trunk pointing to the VCS? (Choose two.)
A. Accept Unsolicited Notification
B. Enable Application Level Authorization
C. Accept Out-of-Dialog REFER
D. Accept Replaces Header
E. Accept Presence Subscription
Answer: A,D
Q7. Scenario
There are two call control systems in this item. The Cisco UCM is controlling the DX650, the Cisco Jabber for Windows Client, and the 9971 Video IP Phone. The Cisco VCS is controlling the SX20, the Cisco TelePresence MCU, and the Cisco Jabber TelePresence for Windows.
Use the exhibits to answer the following questions.
SX20 System Information
DX650 Configuration
MRGL
DP
Locations
AARG
CSS
Movi Failure
Movi Settings
What two issues could be causing the Cisco Jabber Video for TelePresence failure shown in the exhibit? (Choose two.)
A. Incorrect username and password
B. Wrong SIP domain configured.
C. User is not associated with the device.
D. IP or DNS name resolution issue.
E. CSF Device is not registered.
F. IP Phone DN not associated with the user.
Answer: B,D
Q8. Refer to the exhibit.
The Cisco Unified Communications Manager at HQ has been configured for end-to-end RSVP. The Cisco Unified Communications Manager at BR has been configured for local RSVP.
RSVP between the locations assigned to the IP phones and SIP trunks at each site are configured with mandatory RSVP. When a call is placed from the IP phone at the BR site to the IP phone at the HQ site, which statement is true?
A. The Cisco Unified Communications Manager at BR will fall back to local RSVP and place the call. No RSVP end-to-end will occur.
B. RSVP end-to-end will occur.
C. The Cisco Unified Communications Manager at BR will use local RSVP. The HQ Cisco Unified Communications Manager will use end-to-end RSVP.
D. The call will fail.
E. The call will proceed as a normal call with no RSVP reservation.
Answer: A
Q9. Enabling authentication and encryption for CTI, JTAPI, and TAPI applications requires which two tasks? (Choose two.)
A. Enter the encryption key into the application.
B. Set up an IPsec association between the application and Cisco Unified CallManager.
C. Configure related security parameters in the CTI, JTAPI, and TAPI application.
D. Add the application user or end users to the Standard CTI Secure Connection user group, Standard CTI Allow Reception of SRTP Key Material user group, and Standard CTI Enabled user group.
Answer: C,D
Explanation:
Incorrect Answer: A, B You must also add the application users or the end users to the Standard CTI Secure Connection user group in Cisco Unified Communications Manager Administration to enable TLS for the application. After you add the user to this group and install the certificate, the application ensures that the user connects via the TLS port. Link:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/security/8_6_1/secugd/secucti.htm l#wp1166397
Q10. Refer to the exhibit.
Assume that NANP is being used and 9 is used for PSTN access code Long distance national calls are preceded with 1.
How should the HQ Cisco Unified Communications Manager be configured for calls to 3XXX to be sent to the gatekeeper at 1 0 1 6 1 with PSTN backups?
A. Configure a route pattern for 3XXX Assign this route pattern to a route list that points to two route groups The first route group contains the H 225 trunk The second route group contains the MGCP gateway with prefix digits 1 408555 for the outgoing called number.
B. Configure a route pattern for 1#3XXX Assign this route pattern to a route list that points to a route group that lists the H 225 trunk as first choice and the MGCP gateway as a second choice.
C. Configure a route pattern for 4085543XXX. Assign this route pattern to a route list that points to two route groups. The first route group contains the H 226 trunk The second route group contains MGCP gateway.
D. Configure a route pattern for 3XXX Assign this route pattern to a route list that points to two route groups The first route group contains the H 225 trunk The second route group contains MGCP gateway with prefix digits 91 408554 for the called number.
Answer: A
Explanation:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmcfg/b03rtgrp.html #wpxref46617
Q11. Refer to the exhibit.
IT shows an H.323 gateway configuration in a Cisco Unified Communications Manager environment. An inbound PSTN call to this H.323 gateway fails to connect to IP phone extension 2001. The PSTN user hears a reorder tone. Debug isdn q931 on the H.323 gateway shows the correct called-party number as 5015552001. Which two configuration changes can correct this issue? (Choose two.)
A. Add port 1/0:23 to dial-peer voice 123 pots.
B. Ensure that the Significant Digits for inbound calls on the H.323 gateway configuration is 4.
C. Add a voice translation profile to truncate the number from 10 digits to 4 digits. Apply the voice translation profile to the Voice-port. The configuration field "Significant Digits for inbound calls" is left at default (All).
D. Add the command h323-gateway voip id on interface vlan120.
E. Change the destination-pattern on the dial-peer voice 23000 VoIP to 501501? and change the Significant Digits for inbound calls to 4.
Answer: B,E
Explanation:
Incorrect Answer: A, C, D Choose the number of significant digits to collect, from 0 to 32. Cisco Unified Communications Manager counts significant digits from the right (last digit) of the number that is called. Link: http://cisco.biz/en/US/docs/voice_ip_comm/cucmbe/admin/8_6_1/ccmcfg/b06trunk.html
Q12. Which statement is correct about AAR?
A. The end users see, "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.
Answer: A
Explanation:
Incorrect Answer: B, C, D Automated alternate routing (AAR) provides a mechanism to reroute calls through the PSTN or other network by using an alternate number when Cisco Unified Communications Manager blocks a call due to insufficient location bandwidth. With automated alternate routing, the caller does not need to hang up and redial the called party. Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucmbe/admin/8_6_1/ccmcfg/b03aar.ht ml
Q13. If the device pool in the phone record does not match the device pools in the matching subnet, what will the system consider the phone to be?
A. roaming
B. unregistered
C. unknown
D. new device
Answer: A
Q14. Refer to the exhibit.
Which CSS is used at the HQ Cisco Unified Communications Manager to reroute calls via the PSTN when the SAF network is unavailable?
A. the phone device CSS
B. the phone line CSS
C. the phone line/device combined CSS
D. the SAF CSS configured on the CCD requesting service
E. the phone AAR CSS configured at the phone device
F. No special CSS is required. If SAF patterns are accessible, the PSTN reroute is automatic.
Answer: E
Q15. Refer to the exhibit.
All HQ phones are configured to use HQ_MRGL and all BR phones are configured to use BR_MRGL. For the HQ phones always to use the hardware conference bridge as a first choice, which configuration should be implemented?
A. Ensure that both the hardware and software conference bridges are listed in the HQ_MRG. Ensure that the instance ID for the hardware conference bridge is 0.
B. Ensure that both the hardware and software conference bridges are listed in the HQ_MRG. The hardware conference bridge must be configured first.
C. Assign the hardware conference bridge to HQ_MRG. Configure a second HQ_MRG_2 and assign the software conference bridge to it. Add both the HQ_MRG and HQ_MRG_2 to the HQ_MRGL and list the HQ_MRG first.
D. Assign the hardware conference bridge to HQ_MRG. Configure a second HQ_MRG_2 and assign the software conference bridge to it. Configure an additional HQ_MRGL_2. Add the HQ_MRG to HQ_MRGL. Add HQ_MRG_2 to HQ_MRGL_2. The HQ_MRGL should be assigned to the HQ phones. The HQ_MRGL_2 should be assigned to the HQ device pool.
Answer: C
Explanation:
Expiation:
To ensure that the hardware bridge is utilized first with all its resources BEFORE the software bridge is used … you need to have two separate MRG’s and list the hardware MRG 1st in the MRGL …