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Q1. Refer to the exhibit.
The HQ Cisco Unified Communications Manager has been configured for end-to-end
RSVP. The BR Cisco Unified Communications Manager 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 HQ to the BR phone at the BR site, which statement is true?
A. The Cisco Unified Communications Manager at HQ 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 HQ will use end-to-end RSVP. The BR Cisco Unified Communications Manager will use local RSVP.
D. The call will fail.
E. The call will proceed as a normal call with no RSVP reservation.
Answer: D
Explanation:
Incorrect Answer: A, B, C A possible cause is that the same router is being used as the calling and called RSVP agents, and that router is not running the latest IOS version, which supports loopback on RSVP reservation. Make sure that the router is running the latest IOS version. Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a02rsvp.html #wp1155102
Q2. Refer to the exhibit.
Which trunks would be most suitable for Connection Y?
A. an H.225 trunk (gatekeeper-controlled)
B. intercluster trunk (gatekeeper-controlled)
C. a SIP trunk on the U.S. cluster and an intercluster trunk on the remote cluster
D. intercluster trunk (nongatekeeper-controlled)
E. No extra connections are required. As long as IP connectivity exists, you need only configure a route pattern for each site. The Cisco Unified Communications Manager will automatically forward the calls over the WAN if the destination directory number is not registered locally.
Answer: D
Q3. Which action configures AAR to route the calls that have been rejected by the gatekeeper CAC through the PSTN?
A. Configure Cisco IP Phones for AAR.
B. Configure AAR to work with SRST.
C. Configure AAR to work with CTI route points.
D. This configuration is not possible using AAR.
Answer: D
Q4. How do RSVP-enabled locations differ from Cisco Unified Communications Manager locations?
A. RSVP is configured in the ISR independent of Cisco Unified Communications Manager.
B. RSVP enables AAR within Cisco Unified Communications Manager.
C. RSVP is topology aware.
D. RSVP is configured in Cisco Unified Communications Manager independent of the ISR.
Answer: C
Q5. When an H.323 trunk is added for Call Control Discovery, which statement is true?
A. The H.323 trunk is added by selecting Inter-Cluster Trunk (Non-Gatekeeper Controlled) and Device Protocol Inter-Cluster Trunk. The Enable SAF check box should be selected in the trunk configuration.
B. The H.323 trunk is added by selecting Inter-Cluster Trunk (Non-Gatekeeper Controlled) and Device Protocol Inter-Cluster Trunk. The Trunk Service Type should be Call Control Discovery.
C. The H.323 trunk is added by selecting Call Control Discovery Trunk and then selecting
H.323 as the protocol to be used.
D. The H.323 trunk is added by selecting H.323 Trunk, and selecting Inter-Cluster Trunk as the Device Protocol. The destination IP address field is configured as ‘SAF’ to indicate that this trunk is used for SAF.
Answer: A
Explanation:
Reference. Implementing Cisco Unified Communications Manager Part 2 (CIPT2), Chapter3: Implementing Multisite Connections, pg 70-71, Fig 3-14 and Fig 3-15
Q6. 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
Answer: A,D
Explanation:
Incorrect Answer: B, C, E, F For SIP trunks, Cisco Unified Communications Manager supports up to 16 IP addresses for each DNS SRV and up to 10 IP addresses for each DNS host name. The order of the IP addresses depends on the DNS response and may be identical in each DNS query. The OPTIONS request may go to a different set of remote destinations each time if a DNS SRV record (configured on the SIP trunk) resolves to more than 16 IP addresses, or if a host name (configured on the SIP trunk) resolves to more than 10 IP addresses. Thus, the status of a SIP trunk may change because of a change in the way a DNS query gets resolved, not because of any change in the status of any of the remote destinations.
Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a08sip.html
Q7. Which option describes the reason that transcoding resources are added in Cisco Unified Communications Manager?
A. to enable transcoding resources in a Cisco Unified Communications Manager server
B. to enable Cisco Unified Communications Manager to select the optimal single codec for end-to-end calls
C. to enable transcoding resources in Cisco IP Phones
D. to provide transcoding resources in Cisco IOS gateways to Cisco Unified Communications Manager
Answer: D
Q8. 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 and TMS control the Cisco TelePresence MCU, and the Cisco Jabber TelePresence for Windows.
Use the exhibits to answer the following questions.
DP
Locations
CSS
SRST
SRST-BR2 Config
BR2 Config
SRSTPSTNCall
After configuring the CFUR for the directory number that is applied to BR2 phone (+442288224001), the calls fail from the PSTN. Which two of the following configurations if applied to the router, would remedy this situation? (Choose two.)
A. dial-peer voice 1 potsincoming called-number 228822…direct-inward-dialport 0/0/0:15
B. dial-peer voice 1 potsincoming called-number 228822…direct-inward-dialport 0/0/0:13
C. voice translation-rule 1rule 1 /228821 …$/ /+44&/exit!voice translation-profile pstn-intranslate called 1!voice-port 0/0/0:15translation-profile incoming pstn-in
D. voice translation-rule 1rule 1 /228822…$/ /+44&/exit!voice translation-profile pstn-intranslate called 1!voice-port 0/0/0:15translation-profile incoming pstn-in
E. The router does not need to be configured
Answer: A,D
Q9. Which option configures call preservation for H.323-based SRST mode?
A. voice service voip h323 call preserve
B. call preservation not possible with H.323
C. call-manager-fallback preserve-call
D. dial-peer voice 1 voip call preserve
Answer: A
Q10. Which two locations are the best locations that an end user can use to determine if an IP phone is working in SRST mode? (Choose two.)
A. Cisco Unified Communications Manager Administration
B. IP phone display
C. Cisco Unified SRST Router
D. Cisco Unified MGCP Fallback Router
E. physical IP phone settings
Answer: B,E
Explanation:
Incorrect Answer: A, C, D IP Phone display and Physical phone IP settings are two locations were an end user can determine if an IP phone is working in SRST mode. Link: http://my.safaribooksonline.com/book/telephony/1587050757/survivable-remote-site-telephony-srst/529