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2016 Apr 300-075 Study Guide Questions:
Q46. Which two configurable options are available to enable Early Offer for calls over a Cisco Unified Communications Manager SIP trunk? (Choose two.)
A. No Media Termination Point Required
B. Media Termination Point Required
C. Accept Audio Codec Preferences in Received Offer
D. Early Offer support for voice and video calls Mandatory (insert MTP if needed)
E. Use Trusted Relay Point
Q47. When you configure QoS on VCS, which settings do you apply if traffic through the VCS should be tagged with DSCP AF41?
A. Set QoS mode to DiffServ and tag value 32.
B. Set QoS mode to IntServ and tag value to 34.
C. Set QoS mode to DiffServ and tag value 34.
D. Set QoS mode to IntServ and tag value to 32.
E. Set QoS mode to ToS and tag value to 32.
Q48. The administrator at Company X is trying to set up Extension Mobility and has done these steps:
-Set up end users accounts for the users who need to roam
-Set up a device profile for the type of phones users will be allowed to log in Users have reported to the administrator that they are unable to log in to the phones
designated for Extension Mobility. Which two options are the two reasons for this issue? (Choose two.)
A. The user device profile is not associated to the correct end user.
B. The username must be numeric only and must match the DN.
C. The Extension Mobility service has not been enabled under the Cisco Unified Serviceability Page.
D. Extension Mobility has not been enabled under Enterprise Parameters.
E. The user must ensure that their main endpoint is online and registered, otherwise they cannot log in elsewhere.
Q49. What is the standard Layer 3 DSCP media packet value that should be set for Cisco TelePresence endpoints?
A. CS3 (24)
B. EF (46)
C. AF41 (34)
D. CS4 (32)
Q50. Which two statements are true regarding the implementation of globalized call-routing in terms of localized call egress? (Choose two.)
A. Calling-party numbers are routed from the gateway or trunks to phones.
B. Called-party numbers are routed from the gateway or trunks to phones.
C. Calling-party numbers of internal calls are routed from the gateway or trunks.
D. Calling-party calls are routed to the gateway and trunks.
Improved 300-075 free practice exam:
Q51. In a node-specific Service Advertisement Framework forwarder deployment model, what is the maximum number of Service Advertisement Framework forwarders that you can assign to a specific node?
Q52. 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.
Q53. Refer to the exhibit.
To permit three G.729 calls, what should the bandwidth value be for the ip rsvp bandwidth command?
Q54. 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.
Q55. Refer to the exhibit:
The exhibit shows a SAF Forwarder configuration attached to a Cisco Unified Communications Manager.
Which minimum configuration for a Cisco Unified Communications Manager Express SAF Forwarder is needed to establish a SAF neighbor relationship with this SAF Forwarder?
A. router eigrp SAFiservice-family ipv4 autonomous-system 1!topology baseexit-sf-topologyexit-service-familyvoice service safprofile trunkroute 1session protocol sip interface Loopback1 transport tcp port 5060!
B. router eigrp SAF!service-family ipv4 autonomous-system 1!topology baseexit-sf-topologyexit- service-family!voice service safprofile trunk-route 1session protocol sip interface Loopback1 transport tcp port 5060!profile dn-block 1 alias-prefix 1972555pattern 1 type extension 4xxx!profile callcontrol 1dn-servicetrunk-route 1dn-block 1dn-block 2!channel 1 vrouter SAF asystem 1subscribe callcontrol wildcardedpublish callcontrol 1!
C. router eigrp SAF!service-family ipv4 autonomous-system 1!topology baseexit-sf-topologyexit-service-family!
D. None of above configurations contain sufficient information.
Incorrect Answer: A, B, D only following configuration is enough router eigrp SAF service-family ipv4 autonomous-system 1 exit-service-family link:
Breathing 300-075 interactive bootcamp:
Q56. The network administrator of Enterprise X receives reports that at peak hours, some calls between remote offices are not passing through. Investigation shows no connectivity problems. The network administrator wants to estimate the volume of calls being affected by this issue. Which two RTMT counters can give more information on this? (Choose two.)
Q57. 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.
Q58. When an incoming PSTN call arrives at an MGCP gateway, how does the calling number get normalized to a global E.164 number with the + 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 MGCP gateway.
Incorrect Answer: A, B, C, E Configuring calling party normalization alleviates issues with toll bypass where the call is routed to multiple locations over the IP WAN. In addition, it allows Cisco Unified Communications Manager to distinguish the origin of the call to globalize or localize the calling party number for the phone user. Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmfeat/fscallpn.html
Q59. 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?
Q60. Which symbol is required for globalized call routing?
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