Q1. Refer to the exhibit.
The exhibit shows centralized Cisco Unified Communications Manager configuration components for TEHO calls to U.S. area code 408 from the U.K. The PSTN access code for the U.K. is 9 and 001 for international calls to the U.S. Assuming the PSTN does not accept globalized numbers with + prefix. What should the Called Party Transformation Pattern at the U.S. gateway be configured as?
A. \+.! with the following Called Party Transformation:Discard Digits PreDot Prefix Digits Outgoing Calls: +
B. \+1.! with the following Called Party Transformation:Discard Digits PreDot Prefix Digits Outgoing Calls: None
C. \+408.! with the following Called Party Transformation:Discard Digits PreDot Prefix Digits Outgoing Calls: 1
D. \+1408.! with the following Called Party Transformation:Discard Digits PreDot Prefix Digits Outgoing Calls: None
E. \+1.408! with the following Called Party Transformation:Discard Digits PreDot Prefix Digits Outgoing Calls: None
Answer: D
Q2. Which three items must you configure to enable SAF Call Control Discovery? (Choose three.)
A. the SIP or H.323 trunk
B. hosted DN groups
C. hosted DN patterns
D. route patterns
E. a calling search space
F. translation patterns
Answer: A,B,C
Q3. Which commands are needed to configure Cisco Unified Communications Manager Express in SRST mode?
A. telephony-service and srst mode
B. telephony-service and moh
C. call-manager-fallback and srst mode
D. call-manager-fallback and voice-translation
Answer: A
Q4. 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
Q5. Which two statements regarding you configuring a traversal server and traversal client relationship are true? (Choose two.)
A. VCS supports only the H.460.18/19 protocol for H.323 traversal calls.
B. VCS supports either the Assent or the H.460.18/19 protocol for H.323 traversal calls.
C. VCS supports either the Assent or the H.460.18/19 protocol for SIP traversal calls.
D. If the Assent protocol is configured, a TCP/TLS connection is established from the traversal client to the traversal server for SIP signaling.
E. A VCS Expressway located in the public network or DMZ acts as the firewall traversal client.
Answer: B,D
Q6. Refer to the exhibit.
Assume that the HQ phones have access to the HQ partition, and BR phones have access to the BR partition. Which set of implementations would best address the overlapping directory number extensions for intersite (WAN) calling between the HQ site and the BR site?
A. Configure a route pattern 8222.[12]XXX for site HQ, and assign it to partition HQ. Configure the called party DDI of Predot.Configure a route pattern for site BR 8111.[1-3]XXX, and assign it to partition BR. Configure called party DDI Predot.Use the local gateway at each site. Prefix the appropriate site code for the calling number.
B. Configure a single route pattern for both sites 8[12,12,12].[1-32]XXX. Use a route list that contains the local route group for each site. Prefix the appropriate site code for the calling number.
C. Configure a translation pattern 8222.[12]XXX for site HQ, and assign it to partition HQ. Use a CSS that contains the partitions for BR phones.Configure a translation pattern 8111.[1-3]XXX for site BR, and assign it to partition BR. Use a CSS that contains the partitions for HQ phones.For both translation patterns, configure the called party DDI of Predot. Prefix the appropriate site code for the calling number.
D. Configure a translation pattern 8222.[12]XXX for site HQ, and assign it to partition BR. Use a CSS that contains the partitions for HQ phones.Configure a translation pattern 8111.[1-3]XXX for site BR, and assign it to partition HQ. Use a CSS that contains the partitions for BR phones.For both translation patterns, configure the called party DDI of Predot. Prefix the appropriate site code for the calling number.
Answer: C
Q7. What is a prerequisite of AAR deployment?
A. You must have a single distributed call processing deployment.
B. Calls must be manually rerouted through the PSTN or other networks when Cisco Unified Communications Manager blocks a call due to insufficient location bandwidth.
C. Calls must be automatically rerouted through the PSTN or other networks when Cisco Unified Communications Manager blocks a call due to insufficient location bandwidth.
D. Clustering must be implemented over the WAN.
E. You must have a centralized call processing deployment.
Answer: E
Q8. 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
Q9. Which two configurations can you perform to allow Cisco Unified Communications Manager SIP trunks to send an offer in the INVITE? (Choose two.)
A. Enable the Media Termination Point Required option on the SIP trunk.
B. Enable the Early Offer Support for Voice and Video Calls option on the SIP profile.
C. Select the Display IE Delivery check box in the gateway configuration.
D. Select the Enable Inbound FastStart check box on the Cisco Unified Communications Manager servers.
E. Select the SRTP Allowed check box on the SIP trunk.
F. Execute the isdn switch-type primary-ni command globally.
Answer: A,B
Q10. Which statement best describes globalized call routing in Cisco Unified Communications Manager?
A. All incoming calling numbers on the phones are displayed as an E.164 with the + prefix.
B. Call routing is based on numbers represented as an E.164 with the + prefix format.
C. All called numbers sent out to the PSTN are in E.164 with the + prefix format.
D. The CSS of all phones contain partitions assigned to route patterns that are in global format.
E. All phone directory numbers are configured as an E.164 with the + prefix.
Answer: B
Explanation:
Incorrect Answer: A, C, D, E For the destination to be represented in a global form common to all cases, we must adopt a global form of the destination number from which all local forms can be derived. The + sign is the mechanism used by the ITU's E.164 recommendation to represent any PSTN number in a global, unique way. This form is sometimes referred to as a fully qualified PSTN number. Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/dialplan.html#wp1153205