Exam Code: 300 075 pdf (Practice Exam Latest Test Questions VCE PDF)
Exam Name: Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2)
Certification Provider: Cisco
Free Today! Guaranteed Training- Pass cisco 300 075 Exam.
2017 NEW RECOMMEND
Free VCE & PDF File for Cisco 300-075 Real Exam
Pass on Your First TRY 100% Money Back Guarantee Realistic Practice Exam Questions
Q61. Which two features require or may require configuring a SIP trunk? (Choose two.)
A. SIP gateway
B. Call Control Discovery between a Cisco Unified Communications Manager and Cisco Unified Communications Manager Express
C. Cisco Device Mobility
D. Cisco Unified Mobility
E. registering a SIP phone
Incorrect Answer: C, D, E All protocols require that either a signaling interface (trunk) or a gateway be created to accept and originate calls. Device mobility allows Cisco Unified Communications Manager to determine whether the phone is at its home location or at a roaming location. Cisco Unified Mobility gives users the ability to redirect incoming IP calls from Cisco Unified Communications Manager to different designated phones, such as cellular phones. Link:
Q62. 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?
Q63. When considering Extension Mobility, what happens if a user logs into a phone for which the user does not have a user device profile?
A. The phone reboots with an error.
B. If a default device profile for this phone has been configured, it is loaded.
C. The user cannot log in.
D. Another user device profile is loaded.
Q64. Which Cisco IOS command is used to verify that the Cisco Unified Communications Manager Express has registered with the SAF Forwarder?
A. show eigrp service-family ipv4 clients
B. show eigrp address-family ipv4 clients
C. show voice saf dndb all
D. show saf registration
E. show ip saf registration
Incorrect Answer: B, C, D, E show eigrp service-family ipv4 clients Displays information from the EIGRP IPv4 service-family results.
Q65. Which statement is true when considering a Cisco VoIP environment for regional configuration?
A. G.711 requires 128K of bandwidth per call.
B. G.729 requires 24K of bandwidth per call.
C. The default codec does not matter if you have defined a hardware MTP in your Cisco Unified Communications Manager environment.
D. To deploy a Cisco H.323 gatekeeper, you must configure MTP resources on the gatekeeper and only use G.711 between regions.
Q66. Refer to the exhibit.
HQ_MRGL is assigned to the HQ IP phones. BR_MRGL is assigned to the BR IP phones. The remote site BR IP phones support only the G.711 codec. Where should the transcoder reside?
A. The transcoder should reside at the HQ site and assigned to HQ_MRG.
B. The transcoder should reside at the BR site and assigned to BR_MRG.
C. The transcoder should be assigned to its own MRG, which should then be assigned to the default device pool.
D. A transcoder is not needed. The HQ phones will automatically change over to the G.711 codec.
Q67. 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 7965 and 9971 Video IP Phones. The Cisco VCS and TMS control the Cisco TelePresence Conductor, the Cisco TelePresence MCU, and the Cisco Jabber TelePresence for Windows.
Use the exhibits to answer the following questions.
The intercluster URI call routing no longer allows calls between sites. What is the reason why this would happen? (Choose two)
A. Wrong SIP domain configured.
B. User is not associated with the device.
C. IP or DNS name resolution issue.
D. No SIP route patterns for cisco.lab exist.
Q68. Which process can localize a global E.164 with + prefix calling numbers for inbound calls to an IP phone so that users see the calling number in a local format?
A. Calling number localization is done using translation patterns.
B. Calling number localization is done using route patterns.
C. Calling number localization is done by configuring a calling party transformation CSS at the phone.
D. Calling number localization is done by configuring a calling party transformation CSS at the gateway.
E. Calling number localization is done by configuring the phone directory number in a localized format.
Q69. Refer to the exhibit.
IT shows an H.323 gateway configuration in a Cisco Unified Communications Manager environment. An inbound PSTN call to this H.323 gateway fails to connect to IP phone extension 2001. The PSTN user hears a reorder tone. Debug isdn q931 on the H.323 gateway shows the correct called-party number as 5015552001. Which two configuration changes can correct this issue? (Choose two.)
A. Add port 1/0:23 to dial-peer voice 123 pots.
B. Ensure that the Significant Digits for inbound calls on the H.323 gateway configuration is 4.
C. Add a voice translation profile to truncate the number from 10 digits to 4 digits. Apply the voice translation profile to the Voice-port. The configuration field "Significant Digits for inbound calls" is left at default (All).
D. Add the command h323-gateway voip id on interface vlan120.
E. Change the destination-pattern on the dial-peer voice 23000 VoIP to 501501? and change the Significant Digits for inbound calls to 4.
Incorrect Answer: A, C, D Choose the number of significant digits to collect, from 0 to 32. Cisco Unified Communications Manager counts significant digits from the right (last digit) of the number that is called. Link: http://cisco.biz/en/US/docs/voice_ip_comm/cucmbe/admin/8_6_1/ccmcfg/b06trunk.html
Q70. 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