Q1. What happens if location-based CAC is used and there is no bandwidth available when a remote caller is placed on hold?
A. Cisco Unified Communications Manager sends TOH rather than MOH.
B. Cisco Unified Communications Manager terminates the call.
C. Cisco Unified Communications Manager plays default MOH.
D. Cisco Unified Communications Manager attempts to reconnect the call immediately.
Answer: A
Q2. Which component of Cisco Unified Communications Manager is responsible for sending keepalive messages to the Service Advertisement Framework forwarder?
A. Call Control Discovery requesting service
B. Hosted DNs service
C. Service Advertisement Framework client control
D. Cisco Unified Communications Manager database
E. Service Advertisement Framework-enabled trunk
F. gatekeeper
Answer: C
Q3. 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
Q4. Which three devices support the SAF Call Control Discovery protocol? (Choose three.)
A. Cisco Unified Border Element
B. Cisco Unity Connection
C. Cisco IOS Gatekeeper
D. Cisco Catalyst Switch
E. Cisco IOS Gateway
F. Cisco Unified Communications Manager
Answer: A,E,F
Q5. 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. Which statement is true?
A. The AAR group system must be configured on the device configuration of the phones.
B. The AAR group system must be configured on the line configuration of the phones.
C. The single AAR group system cannot be used. A second AAR group must be configured in order to have source and destination AAR groups.
D. The AAR group system must be configured under the AAR service parameters.
Answer: B
Q6. Refer to the exhibit.
How many calls are permitted by the RSVP configuration?
A. one G.711 call
B. two G.729 calls
C. one G.729 call and one G.711 call
D. eight G.729 calls
E. four G.729 calls
Answer: B
Explanation:
Incorrect Answer: A, C, D, E In performing location bandwidth calculations for purposes of call admission control, Cisco Unified Communications Manager assumes that each call stream consumes the following amount of bandwidth:
.G.711 call uses 80 kb/s.
.G.722 call uses 80 kb/s.
.G.723 call uses 24 kb/s.
.G.728 call uses 26.66 kb/s.
.G.729 call uses 24 kb/s.
Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a02cac.html #wpxref28640
Q7. Which two options are effective mechanisms to restrict the maximum number of voice calls on a WAN link? (Choose two.)
A. Configure a gatekeeper with an SIP trunk.
B. Configure a gatekeeper and a gatekeeper-controlled trunk in Cisco Unified Communications Manager with bandwidth control.
C. Configure Cisco Unified Communications Manager regions.
D. Configure Cisco Unified Communications Manager locations.
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. How does the system intelligently shift call processing upon restoration of WAN connectivity?
A. automatically back to the primary Cisco Unified Communications Manager cluster
B. manually back to the primary Cisco Unified Communications Manager cluster
C. automatically back to the secondary Cisco Unified Communications Manager cluster
D. manually back to the secondary Cisco Unified Communications Manager cluster
Answer: A
Q10. Refer to the following exhibits.
Users in the U.S dial Germany by calling 9011 49 followed by the remaining digits. What would be the most suitable configuration for Connection X?
A. Configure a SIP trunk to 10.140.1.1 and a SIP route pattern +49T in Cisco Unified Communications Manager.
B. Configure a SIP trunk to the Cisco Unified Border Element and route pattern +49T in Cisco Unified Communications Manager.
C. configure a SIP trunk to the Cisco Unified Border Element. Configure a translation pattern for 9011.49T using DDI Predot prefix + and CSS to point to a route pattern partition \+! which uses the SIP trunk.
D. Configure a SIP trunk to the ITSP. Configure a translation pattern for 9011.49T using DDI predot prefix + and CSS to point to a route pattern partition \+! which uses the SIP trunk.
Answer: C
Explanation:
Incorrect Answer: A, B, D SIP trunks for public switched telephone network (PSTN) access are an important new access method for business collaboration. Service providers throughout the world offer SIP trunking as an alternative to traditional TDM (T1/E1) connections. A discard digits instruction (DDI) removes a portion of the dialed digit string before passing the number on to the adjacent system. A DDI must remove portions of the digit string, for example, when an external access code is needed to route the call to the PSTN, but the PSTN switch does not expect that access code.
Link: https://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a03rp.html