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Q121. 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. 

DNS Servers 

Device Pool 





Speed Dial 

SIP Trunk 

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. 

Answer: C,D 

Q122. Which three options describe the main functions of SAF Clients? (Choose three.) 

A. registering the router as a client with the SAF network B. providing publishing services to the SAF network 

C. subscribing to SAF network services 

D. registering Cisco Unified Communications Manager subscribers with the publisher 

E. starting Cisco Unified Communications Manager services throughout the cluster 

F. integrating with Cisco IM and Presence for additional services 

Answer: A,B,C 

Q123. Refer to the exhibit. 

Which statement about the configuration between the Default and BR regions is true? 

A. Calls between the two regions can use either 64 kbps or 8 kbps. 

B. Calls between the two regions can use only the G.729 codec. 

C. Only 64 kbps will be used between the two regions because the link is "lossy". 

D. Both codecs can be used depending on the loss statistics of the link. When lossy conditions are high, the G.711 codec will be used. 


Q124. 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 

Answer: A,D 

Q125. A voice-mail product that supports only the G.711 codec is installed in headquarters. 

Which action allows branch Cisco IP phones to function with voice mail while using only the G.729 codec over the WAN link to headquarters? 

A. Configure Cisco Unified Communications Manager regions. 

B. Configure transcoding within Cisco Unified Communications Manager. 

C. Configure transcoding resources in Cisco IOS and assign to the MRGL of Cisco IP phones. 

D. Configure transcoder resources in the branch Cisco IP phones. 


Q126. Which method can be used to address variable-length dial plans? 

A. Overlap sending and receiving. 

B. Add a prefix for all calls that are longer than 10-digits long 

C. Use nested translation patterns to eliminate inter-digit timeout 

D. Use the @macro on the route pattern 

E. Use MGCP gateways, which support variable-length dial plans 



Incorrect Answer: B, C, D, E If the dial plan contains overlapping patterns, Cisco Unified Communications Manager does not route the call until the interdigit timer expires (even if it is possible to dial a sequence of digits to choose a current match). Check this check box to interrupt interdigit timing when Cisco Unified Communications Manager must route a call immediately. By default, the Urgent Priority check box displays as checked. Unless your dial plan contains overlapping patterns or variable length patterns that contain!, Cisco recommends that you do not uncheck the check box. Link: 

Q127. Refer to the exhibit. 

When a user presses a speed dial to +442079460255 when the SAF network is down, which event should occur? 

A. The call will reroute via the PSTN with the constructed PSTN number as 442079460255. 

B. The call will reroute via the PSTN with the constructed PSTN number as +442079460255. 

C. The call will reroute via the PSTN with the constructed PSTN number as 00442079460255. 

D. The call will fail because the ToDID is 0:. 

E. The call will fail because the called number will be 2079460255. 


Q128. Which option describes a function of SIP preconditions? 

A. SIP preconditions enable end-to-end RSVP over an SIP trunk. 

B. SIP preconditions enable RSVP between Cisco IP Phones. 

C. SIP preconditions can be enabled in a gatekeeper. 

D. SIP preconditions enable end-to-end RSVP for calls through the PSTN. 


Q129. When an external call is placed from Ajax, they would like the ANI that is sent to the PSTN to be the main number, not the extension. For domestic calls, they would like 10 digits sent; for international calls, they would like to send the country code 1 and the 10 digits. How can this be accomplished? 

A. Add a translation pattern to the dial peers in the gateway that adds the appropriate digits to the outgoing ANI. 

B. In the external call route patterns, set the external phone number mask to the main number. Use 10 digits in the domestic route pattern and 1 followed by the main number digits in the international route patterns. 

C. Use a calling party transform mask for each route group in the corresponding route list configuration. Set the explicit 10-digit main number for domestic calls and 1 followed by the main number for the international route patterns. 

D. In the directory number configurations, set the prefix digits field to the country code and the 10 digits of the main number. This will be truncated to the 10-digit number for domestic calls and sent out in its entirety for international calls. 



Incorrect Answer: A, B, D calling party transformation mask value is Valid entries for the NANP include the digits 0 through 9; the wildcard characters X, asterisk (*), and octothorpe (#); and the international escape character +. Link: ml 

Q130. 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.