Act now and download your Cisco 300 075 pdf test today! Do not waste time for the worthless Cisco 300 075 vce tutorials. Download Regenerate Cisco Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2) exam with real questions and answers and begin to learn Cisco 300 075 ciptv2 with a classic professional.
Q111. Which action configures transcoding resources in Cisco Unified Communications Manager to function with branch office Cisco IP Phones?
A. Configure the branch office IP phones with CSS and partitions.
B. Configure the branch office IP phones with MRGs and MRGLs.
C. Configure the branch office IP phones with regions.
D. Configure the branch office IP phones with locations.
Q112. 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
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.
Q113. Which bandwidth amounts are correct for configuring locations?
A. 8 kb/s for G.729, 64 kb/s for G.711, and 64 kb/s for G.722
B. 8 kb/s for G.729, 64 kb/s for G.711, and 16 kb/s for G.722
C. 64 kb/s for G.729, 64 kb/s for G.711, and 64 kb/s for G.722
D. 8 kb/s for G.729, 8 kb/s for G.711, and 8 kb/s for G.722
Q114. When you use the Query wizard to configure the trace and log central feature to collect install logs, if you have servers in a cluster in a different time zone, which time is used?
A. TLC adjusts the time change appropriately.
B. TLC uses its local time for all systems.
C. TLC queries for the time zone as part of configuration.
D. TLC produces an error and must be run remotely.
Q115. What is the difference between an H.323 gateway and a SIP gateway?
A. An H.323 gateway requires that dial peers be configured before PSTN calls can be placed and received. The SIP gateway requires no dial peers.
B. The H.323 gateway can be added in Cisco Unified Communications Manager under gateway type as H.323 Gateway. The SIP gateway can connect to Cisco Unified Communications Manager only through a SIP trunk.
C. A SIP gateway requires a call agent for PSTN calls to be placed and received. An H.323 gateway does not require a call agent for PSTN calls to be placed and received.
D. An H.323 gateway can register with Cisco Unified Communications Manager. A SIP gateway will show status of "Unknown".
E. The H.323 gateway must be configured in Cisco Unified Communications Manager using a valid IP address on the gateway. The SIP gateway must be configured in Cisco Unified Communications Manager using the domain name.
Q116. 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
Q117. Which ability does the Survivable Remote Site Telephony feature provide?
A. a means to allow the local site to continue to send and receive calls in the event of a WAN failure
B. a means to route calls on-net through other sites during high utilization periods
C. a method that allows for backup calls in the event that your gateway fails
D. the ability to force a call out of a certain trunk when the Cisco Unified Communications Manager is being upgraded
Q118. When an incoming PSTN call arrives at an H.323 gateway, how does the calling number get normalized to a global E.164 number with + prefix in Cisco Unified Communications Manager?
A. Normalization is done using translation patterns.
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 H.323 gateway.
Q119. Which two options are valid service parameter settings that are used to set up proper video QoS behavior across the Cisco Unified Communications Manager infrastructure? (Choose two.)
A. DSCP for Video Calls when RSVP Fails
B. Default Intraregion Min Video Call Bit Rate (Includes Audio)
C. Default Interregion Max Video Call Bit Rate (Includes Audio)
D. DSCP for Video Signaling
E. DSCP for Video Signaling when RSVP Fails
Q120. 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.
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.
To know more about the 300-075, click here.