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Q41. Refer to the exhibit 

When the Cisco Unified Communications Manager advertises the Hosted DN Pattern, which pattern would be advertised? 

A. 2XXX and the T0D1D will be 0:+498950555 

B. 2XXX and the ToDID will be 0:+4989531 21 

C. 4989S05552XXX and the ToDiD will be 0: 

D. + 4989631 21 2XXX and the ToDiD will be 0: 

E. Both +4989505552XXXand +4989531 21 2XXX will be advertised with ToDID of 0: 

Answer:

Explanation: 

Incorrect Answer: B, C, D, E PSTN failover prepend digit is +498950555 Link: 

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_0_2/ccmfeat/fscallcontrol discovery.html 


Q42. Which action configures phones in site A to use G.711 to site B but to use G.729 to site C? 

A. Configure Cisco Unified Communications Manager regions. 

B. Configure Cisco Unified Communications Manager locations. 

C. Configure transcoder resources in Cisco Unified Communications Manager. 

D. Configure a gatekeeper. 

Answer:


Q43. 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) 

Answer:


Q44. Which action configures the registration of transcoder resources? 

A. Cisco IOS software registers transcoder resources with SIP. 

B. Cisco IOS software registers transcoder resources with SCCP. 

C. Cisco IOS software registers transcoder resources with H.323. 

D. Cisco IOS software does not register transcoder resources. 

Answer:


Q45. On which Cisco Unified Communications Manager configuration parameter does the CODEC that a Cisco IP Phone uses for a call depend? 

A. enterprise parameters 

B. media resources 

C. physical location 

D. region 

E. location 

Answer:


Q46. 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? 

A. G.722 

B. G.711 

C. G.729 

D. iSAC 

E. GSM-FR 

F. iLBC 

Answer:


Q47. 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 


Q48. Which statement about SIP precondition is most correct? 

A. When configuring SIP precondition, the SIP trunk must have access to an RSVP agent. 

B. When configuring SIP precondition, the IP phones must have access to an RSVP agent. 

C. When configuring SIP precondition, the IP phones and SIP trunk must have access to an RSVP agent. 

D. RSVP agents are only required for the IP phones. SIP trunks require RSVP agents only when fall back to local RSVP is configured. 

E. SIP trunk will always require RSVP agents regardless of what RSVP type is configured. 

Answer:


Q49. You are deploying a Cisco Unified Communications Manager solution with MGCP gateways at multiple locations. Which firewall and ACL configuration must you perform to allow the MCGP gateways to function correctly? 

A. Allow access to TCP port 2428. 

B. Block TCP port 1720. 

C. Open access to all TCP and UDP ports. 

D. Allow access to TCP port 1720. 

E. Block access to TCP ports 2427 and 2428. 

Answer:


Q50. In what Cisco solution is Simple Network-Enabled Auto Provision technology used? 

A. Cisco Unified Gateway Duplication 

B. Cisco Unified CallManager Redundancy 

C. Cisco Unified SRST 

D. Cisco Unified Call Survivability 

Answer:

Explanation: 

Incorrect Answer: A, B, D When the system automatically detects a failure, Cisco Unified SRST uses Simple Network Auto Provisioning (SNAP) technology to auto-configure a branch office router to provide call processing for the Cisco Unified IP phones that are registered with the router Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmesr st.html