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Q46. 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 

Q47. Which E.164 transformation pattern represents phone numbers in Germany? 

A. \+49.! 

B. 49.! 

C. \49.! 

D. \+49.X 

Answer: A 

Q48. Which two entities could be represented by device mobility groups? (Choose two.) 

A. countries 

B. regions 

C. directory numbers 

D. transcoders 

Answer: A,B 

Q49. Which two configurations can you perform to allow Cisco Unified Communications Manager SIP trunks to send an offer in the INVITE? (Choose two.) 

A. Enable the Media Termination Point Required option on the SIP trunk. 

B. Enable the Early Offer Support for Voice and Video Calls option on the SIP profile. 

C. Select the Display IE Delivery check box in the gateway configuration. 

D. Select the Enable Inbound FastStart check box on the Cisco Unified Communications Manager servers. 

E. Select the SRTP Allowed check box on the SIP trunk. 

F. Execute the isdn switch-type primary-ni command globally. 

Answer: A,B 

Q50. When Cisco Extension Mobility is implemented, how is the audio source for the MOH selected? 

A. The audio source that is configured at the user device profile is selected. 

B. The audio source that is configured at the home phone of the user is selected. 

C. The audio source that is configured at the physical phone used for the Cisco Extension Mobility login is selected. 

D. The audio source that is configured in the IP Voice Media Streaming parameters is selected. 

Answer: A 

Explanation: 

Incorrect Answer: B, C, D To specify the audio source that plays when a user initiates a hold action, choose an audio source from the User Hold MOH Audio Source drop-down list box from device profile configuration settings. Link: http://cisco.biz/en/US/docs/voice_ip_comm/cucmbe/admin/8_6_1/ccmcfg/b06dvprf.html 

Q51. Assume that the Cisco IOS SAF Forwarder is configured correctly. Which minimum configurations on Cisco Unified Communications Manager are needed for the SAF registration to take place? 

A. SAF Trunk, SAF Security Profile, SAF Forwarder, and CCD Advertising Service 

B. SAF Trunk, SAF Security Profile, SAF Forwarder, and CCD Requesting Service 

C. SAF Trunk, SAF Security Profile, SAF Forwarder, CCD Requesting Service, and CCD Advertising Service 

D. SAF Trunk, SAF Security Profile, and SAF Forwarder 

E. SAF Trunk, CCD Requesting Service, and CCD Advertising Service 

Answer: B 

Q52. Company X has three locations connected via a low bandwidth WAN. Which two configurations are required in the Cisco Unified Communications Manager regions to provide the most suitable use of bandwidth while preserving the call quality? (Choose two.) 

A. g729 codec for intraregion calling 

B. g722/g711 codec for interregion calling 

C. g729 codec for interregion calling 

D. g722/g711 for intraregion calling 

E. g729 codec for all calling 

F. g722/g711 codec for all calling 

Answer: C,D 

Q53. Which three configuration settings are included in a default region configuration? (Choose three.) 

A. Audio Codec 

B. Video Call Bandwidth 

C. Link Loss Type 

D. Location Description 

E. Immersive Bandwidth 

F. Real Time Protocol 

Answer: A,B,C 

Q54. Refer to the exhibit. 

The HQ Cisco Unified Communications Manager has been configured for end-to-end 

RSVP. The BR Cisco Unified Communications Manager has been configured for local 

RSVP. 

RSVP between the locations assigned to the IP phones and SIP trunks at each site are configured with mandatory RSVP. When a call is placed from the IP phone at HQ to the BR phone at the BR site, which statement is true? 

A. The Cisco Unified Communications Manager at HQ will fall back to local RSVP and place the call. No RSVP end-to-end will occur. 

B. RSVP end-to-end will occur. 

C. The Cisco Unified Communications Manager at HQ will use end-to-end RSVP. The BR Cisco Unified Communications Manager will use local RSVP. 

D. The call will fail. 

E. The call will proceed as a normal call with no RSVP reservation. 

Answer: D 

Explanation: 

Incorrect Answer: A, B, C A possible cause is that the same router is being used as the calling and called RSVP agents, and that router is not running the latest IOS version, which supports loopback on RSVP reservation. Make sure that the router is running the latest IOS version. Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a02rsvp.html #wp1155102 

Q55. 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 adding SRST functionality the SRST does not work. After reviewing the exhibits, which of the following reasons could be causing this failure? 

A. Device Pool cannot be default. 

B. The Cisco UCM is pointing to the wrong IPv4 address of the BR router. 

C. The router does not support SRST. 

D. The SRST enabled router is not configured correctly. 

Answer: A 

Q56. Refer to the exhibit. A user in RTP calls a phone in San Jose during congestion with Call Forward No Bandwidth (CFNB) configured to reach cell phone 4085550150. The user in RTP sees the message "Not Enough Bandwidth" on their phone and hears a fast busy tone. Which two conditions can correct this issue? (Choose two.) 

A. The calling phone (RTP) needs to have AAR Group value of AAR under the AAR Settings. 

B. The called phone (San Jose) needs to have AAR Group value of AAR under the AAR Settings. 

C. The calling phone (RTP) needs to have the AAR destination mask of 914085550150 configured under the AAR Settings. 

D. The calling phone (RTP) needs to have the AAR destination mask of 4085550150 configured under the AAR Settings. 

E. The called phone (San Jose) needs to have the AAR destination mask of 914085550150 configured under the AAR Settings. 

F. The called phone (San Jose) needs to have the AAR destination mask of 4085550150 configured under the AAR Settings. 

Answer: B,F 

Explanation: 

Incorrect Answer: A, C, D, E Automated alternate routing (AAR) provides a mechanism to reroute calls through the PSTN or other network by using an alternate number when Cisco Unified Communications Manager blocks a call due to insufficient location bandwidth. With automated alternate routing, the caller does not need to hang up and redial the called party. Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucmbe/admin/8_6_1/ccmcfg/b03aar.ht ml 

Q57. 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 

A third collaboration call fails between the backbone site and the HQ site. After reviewing the exhibits, which of the following reasons could be causing this failure? 

A. Not enough bandwidth has been allocated. 

B. Device Pool. 

C. Location. 

D. The pipe is not functioning. 

Answer: A 

Explanation: 

Based on the exhibit, each call is limited to no more than 128kbps per call, but the total available bandwidth is set to 256 kbps. This will allow the first to calls to go through, but there will be no more available bandwidth for the third call. 

Q58. Which two are gatekeeper-controlled trunk options that support gatekeeper call administration control? (Choose two.) 

A. H.323 

B. H.245 

C. H.225 

D. intercluster 

E. intracluster 

Answer: C,D 

Q59. Which three options are supplementary services that are affected by MTP? (Choose three.) 

A. Call Hold 

B. Call Transfer 

C. Call Park 

D. Call Pickup 

E. Speed Dial 

F. Call Back 

Answer: A,B,C 

Q60. Refer to the exhibit. 

How does the Cisco Unified Communications Manager advertise dn-block 2? 

A. 14087071222 with number type international 

B. +14087071222 with number type international 

C. +14087071222 

D. 14087071222 

Answer: C