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Q31. What impact do roaming-sensitive settings and Device Mobility settings have on call routing? 

A. Device Mobility settings have no impact on call routing, but roaming-sensitive settings modify the AAR group, AAR CSS, and device CSS. 

B. Device Mobility settings modify the device CSS and the roaming-sensitive settings modify the AAR group and AAR CSS. 

C. Device Mobility settings modify the AAR group and the AAR CSS, the roaming-sensitive settings modify the device CSS. 

D. Roaming-sensitive settings are settings that do not have an impact on call routing. Device Mobility settings, on the other hand, may have an impact on call routing because they modify the device CSS, AAR group, and AAR CSS. 

Answer: D 


Q32. Which two entities could be represented by device mobility groups? (Choose two.) 

A. countries 

B. regions 

C. directory numbers 

D. transcoders 

Answer: A,B 


Q33. 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. 

Answer: B 


Q34. Which device is needed to integrate H.320 into the Cisco video solution? 

A. video gateway 

B. MGCP gateway 

C. H.323 gatekeeper 

D. MCU 

Answer: C 

Explanation: 

Incorrect Answer: A, B, D As with H.323 MCUs, H.320 gateways are provisioned in Cisco Unified CallManager as H.323 gateways, and then route patterns are configured to extend calls to these devices. 

Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/4x/42video.html#wp1046523 


Q35. 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. 

Answer: C 


Improve 300-075 rapidshare:

Q36. Scenario 

There are two call control systems in this item. The Cisco UCM is controlling the Cisco Jabber for Windows Client, and the 7965 and 9971 Video IP Phone. The Cisco VCS is controlling the SX20, the Cisco TelePresence MCU, and the Cisco Jabber TelePresence for Windows. 

Use the exhibits to answer the following questions. 

DP 

Locations 

CSS 

Movi Failure 

Movi Settings 

CIPTV Topo 

Subzone 

Links 

Pipe 

Both of the Cisco TelePresence Video for Windows clients are able to log into the server but can’t make any calls. After reviewing the exhibits, which of the following reasons could be causing this failure? 

A. Wrong username and/or password. 

B. Wrong SIP domain name. 

C. The TMSPE is not working. 

D. The bandwidth is incorrectly configured. 

Answer: D 


Q37. If your IP telephony administrator asks you to configure a local zone for your dial plan to control the volume of calls between two end points in a centralized multisite environment, which two types of Call Admission Control can be implemented? (Choose two.) 

A. locations based 

B. automated alternate routing 

C. gatekeeper based 

D. SRST 

E. Cisco Unified Communications Manager based 

Answer: A,B 

Explanation: 

Incorrect Answer: C, D, E Location-based call admission control (CAC) manages WAN link bandwidth in Cisco Unified Communications Manager. Automated alternate routing (AAR) provides a mechanism to reroute calls through the PSTN or other network by using an alternate number. Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a02cac.html #wp1067747 


Q38. 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 

Answer: B,D 


Q39. 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. 

Answer: A 


Q40. 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 9971 Video IP Phone. The Cisco VCS and TMS control the Cisco TelePresence MCU, and the Cisco Jabber TelePresence for Windows. 

Use the exhibits to answer the following questions. 

DP 

Locations 

CSS 

SRST 

SRST-BR2 Config 

BR2 Config 

SRSTPSTNCall 

After configuring the CFUR for the directory number that is applied to BR2 phone (+442288224001), the calls fail from the PSTN. Which two of the following configurations if applied to the router, would remedy this situation? (Choose two.) 

A. dial-peer voice 1 potsincoming called-number 228822…direct-inward-dialport 0/0/0:15 

B. dial-peer voice 1 potsincoming called-number 228822…direct-inward-dialport 0/0/0:13 

C. voice translation-rule 1rule 1 /228821 …$/ /+44&/exit!voice translation-profile pstn-intranslate called 1!voice-port 0/0/0:15translation-profile incoming pstn-in 

D. voice translation-rule 1rule 1 /228822…$/ /+44&/exit!voice translation-profile pstn-intranslate called 1!voice-port 0/0/0:15translation-profile incoming pstn-in 

E. The router does not need to be configured 

Answer: A,D