300-075 paper（41 to 50） for IT learners: Mar 2017 Edition
Exam Code: 300-075 (Practice Exam Latest Test Questions VCE PDF)
Exam Name: Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2)
Certification Provider: Cisco
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2017 Mar 300-075 question
Q41. When a call is made from a video endpoint to a Cisco TelePresence EX90 that is registered to a Cisco VCS Control, which portion of the destination URI is the first match that is attempted?
A. the full URI, including the domain portion
B. the destination alias, without the domain portion
C. the E.164 number that is assigned to the Cisco TelePresence EX90
D. the directory number that is assigned to the Cisco TelePresence EX90
Q42. Refer to the exhibit.
What should the destination IP address be configured as on the HQ and BR1 SIP trunks?
A. The HQ SIP trunk destination IP address should be 10.1.6.10. The BR1 SIP trunk destination IP address should be 10.1.5.10.
B. The destination IP address is not configurable. Each cluster will learn about the remote trunk IP address through SAF learned routes.
C. The destination IP address will be learned automatically and configured on the SIP trunks after the Cisco Unified Communications Managers discover themselves.
D. The HQ SIP trunk destination IP address should be the HQ SAF Forwarder IP address. The BR1 SIP trunk destination IP address should be the BR1 SAF Forwarder IP address.
Incorrect Answer: A, C, D
The gatekeeper changes the IP address of this remote device dynamically to reflect the IP address of the remote device.
Q43. When an incoming PSTN call arrives at an MGCP 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 MGCP gateway.
B. Normalization is done using route patterns.
C. Normalization is done using the gateway incoming called party prefixes based on number type.
D. Normalization is done using the gateway incoming calling party prefixes based on number type.
E. Normalization is achieved by local route group that is assigned to the MGCP gateway.
Q44. Which remote-site redundancy technology fails over to POTS dial peers from the Cisco Unified Communications Manager dial plan during a WAN failure?
A. MGCP fallback
B. H.323 fallback
C. SCCP fallback
D. SIP fallback
Q45. 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 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.
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
Regenerate 300-075 download:
Q46. 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.
Q47. What is the default DSCP/PHB for TelePresence video conferencing packets in Cisco Unified Communications Manager?
Q48. 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
Incorrect Answer: A, C, D A 384-kb/s video call may comprise G.711 at 64 kb/s (for audio) plus 320 kb/s (for video). This sum does not include overhead. If the audio codec for a video call is G.729 (at 24 kb/s), the video rate increases to maintain a total bandwidth of 384 kb/s. Link:
Q49. An update of the configuration using the Cisco CTL client not needed when _______.
A. a Cisco Unified CallManager has been removed
B. an LSC of the IP phone is upgraded
C. a security token is added to the system
D. an IP address of the Cisco TFTP server has been changed
Incorrect Answer: A, C, D The CTL file contains entries for the following servers or security tokens:
. System Administrator Security Token (SAST)
. Cisco CallManager and Cisco TFTP services that are running on the same server
. Certificate Authority Proxy Function (CAPF) .
. TFTP server(s)
. ASA firewall
Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/security/8_6_1/secugd/secuauth.h tml#wp1028878
Q50. When Cisco Extension Mobility is implemented, which CSS is used for calling privileges?
A. The user device profile line CSS combined with the device CSS of the physical phone used to log in the extension mobility user.
B. The user device profile device CSS combined with the line CSS of the physical phone used to log in the extension mobility user.
C. Only the user device profile device CSS is used.
D. The combined line/device CSS of the physical phone is used to log in the extension mobility user.
E. The combined line/device CSS of the user device profile.