Pinpoint of 300-075 free practice exam materials and dumps for Cisco certification for customers, Real Success Guaranteed with Updated 300-075 pdf dumps vce Materials. 100% PASS Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2) exam Today!
2016 Jun 300-075 Study Guide Questions:
Q91. You are the Cisco Unified Communications Manager in Certpaper.com. You use a remote site MGCP gateway to provide redundancy when connectivity to the central Cisco Unified Communications Manager cluster is lost. How to enable IP phones to establish calls to the PSTN when they have registered with the gateway? (Choose three.)
A. POTS dial peers must be added to the gateway to route calls from the IP phones to the PSTN.
B. The default service must be enabled globally.
C. The command ccm-manager mgcp-fallback must be configured.
D. COR needs to be configured to disallow outbound calls.
Incorrect Answer: D Class of restriction: Cisco Unified Communications Manager Business Edition 3000 supports class of service (CoS) with respect to geographic reach as follows:
– Emergency services
. Call waiting
. Default ringtones
. Speed dials: Single-button, not BLF
Q92. What component acts as a user agent for both ends of a SIP call while Cisco Unified SIP SRST is providing failover during a WAN outage?
B. SIP server
C. SIP proxy
D. SIP SRST router
E. SIP registrar
Q93. Which of the following are two functions that ensure that the telephony capabilities stay operational in the remote location Cisco Unified SRST router? (Choose two)
A. Automatically detecting a failure in the network.
B. Initiating a process to provide call-processing backup redundancy.
C. Notifying the administrator of an issue for manual intervention.
D. Proactively repairing issues in the voice network.
Abreast of the times pass4sure 350-018 pdf:
Q94. 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
Q95. Which statement about H.323 Gatekeeper Call Admission Control is true?
A. Gatekeeper Call Admission Control applies to centralized Cisco Unified Communications deployments that use locations based Call Admission Control.
B. Gatekeeper Call Admission Control applies to distributed Cisco Unified Communications deployments.
C. Gatekeeper Call Admission Control applies only to distributed Cisco Unified Communications Express deployments.
D. Gatekeeper Call Admission Control setting is configured in Cisco Unified Communications Manager.
Incorrect Answer: A, C, D in distributed call processing deployments on a simple hub-and-spoke topology, you can implement call admission control with a Cisco IOS gatekeeper. In this design, the call processing agent (which could be a Unified CM cluster, Cisco Unified Communications Manager Express (Unified CME), or an H.323 gateway) registers with the Cisco IOS gatekeeper and queries it each time the agent wants to place an IP WAN call. Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/cac.html#wp1044743
Q96. You have been asked to deploy Cisco Extension Mobility Cross Cluster for a distributed call processing environment. During the initial extension mobility login request, how does the visiting cluster determine if the user is a local user or a remote user?
A. by using a third-party automatic provisioning tool to verify user ID
B. by broadcasting a request to all clusters to verify the user type
C. from user IDs that are created by default when the user logs in
D. by using Extension Mobility Cross Cluster Session Initiation Protocol (SIP) trunks
E. by verifying against the local database
F. by verifying the visiting Trivial File Transfer Protocol
Simulation cisco 350-018 exam:
Q97. You are deploying a remote office setup that connects with Cisco Unity Communications Manager at a hub location. You have an available dedicated bandwidth of 20% from the 2-Mb/s WAN circuit for VoIP that supports a maximum of 17 calls. Which codec do you configure in Cisco Unity Communications Manager to achieve this?
Q98. How is a SIP trunk in Cisco Unified Communications Manager configured for SIP precondition?
A. The configuration is done by selecting a SIP precondition trunk for trunk type.
B. The configuration is automatically selected when RSVP is enabled for the location assigned to the trunk.
C. SIP precondition is configured by selecting E2E for RSVP over SIP on the default SIP profile assigned to the SIP trunk.
D. SIP precondition is configured by configuring a new SIP profile and selecting E2E for RSVP over SIP. The new SIP profile must then be assigned to the SIP trunk.
Q99. 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.
Reference. Implementing Cisco Unified Communications Manager Part 2 (CIPT2), Chapter3: Implementing Multisite Connections, pg 70-71, Fig 3-14 and Fig 3-15
Q100. The corporate WAN has been extended to two newly acquired sites and it includes gatekeeper support. Each site has a Cisco CallManager and an H.323 gateway that allows connection to the PSTN. Which connection method is best for these two new customers?
A. H.225 trunk (gatekeeper-controlled)
B. intercluster trunk (non-gatekeeper controlled)
C. SIP trunk
D. intercluster trunk (gatekeeper-controlled)