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Q131. Refer to the exhibit. 

How many calls are permitted by the RSVP configuration? 

A. one G.711 call 

B. two G.729 calls 

C. one G.729 call and one G.711 call 

D. eight G.729 calls 

E. four G.729 calls 

Answer:

Explanation: 

Incorrect Answer: A, C, D, E In performing location bandwidth calculations for purposes of call admission control, Cisco Unified Communications Manager assumes that each call stream consumes the following amount of bandwidth: 

.G.711 call uses 80 kb/s. 

.G.722 call uses 80 kb/s. 

.G.723 call uses 24 kb/s. 

.G.728 call uses 26.66 kb/s. 

.G.729 call uses 24 kb/s. 

Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a02cac.html #wpxref28640 


Q132. 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:


Q133. Which two options enable routers to provide basic call handling support for Cisco Unified IP Phones if they lose connection to all Cisco Unified Communications Manager systems? 

(Choose two.) 

A. SCCP fallback 

B. Cisco Unified Survivable Remote Site Telephony 

C. Cisco Unified Communications Manager Express 

D. MGCP fallback 

E. Cisco Unified Communications Manager Express in SRST mode 

Answer: B,E 


Q134. Which system configuration is used to set a restriction on audio bandwidth? 

A. region 

B. location 

C. physical location 

D. licensing 

Answer:


Q135. Which commands are needed to configure Cisco Unified Communications Manager Express in SRST mode? 

A. telephony-service and srst mode 

B. telephony-service and moh 

C. call-manager-fallback and srst mode 

D. call-manager-fallback and voice-translation 

Answer:


Q136. With Cisco Extension Mobility, when a user logs in to a phone type which has no user device profile, what will happen to the phone? 

A. The phone takes on the default clusterwide device profile. 

B. The phone creates a new device profile automatically. 

C. The phone immediately logs the user off. 

D. The phone crashes and reboots. 

Answer:


Q137. When multiple Cisco Extension Mobility profiles exist, which actions take place when a user tries to log in to Cisco Extension Mobility? 

A. The login will fail because only a single Cisco Extension Mobility profile is allowed. 

B. The user must select the desired profile. 

C. The user must login to both profiles in the order they are presented. 

D. The user may login to both profiles in any order. 

E. Login will only be allowed to multiple profiles if the service parameter Allow Multiple Logins is enabled. 

Answer:

Explanation: 

Incorrect Answer: A, C, D, E Users access Cisco Extension Mobility by pressing the Services or Applications button on a Cisco Unified IP Phone and then entering login information in the form of a Cisco Unified Communications Manager UserID and a Personal Identification Number (PIN). If a user has more than one user device profile, a prompt displays on the phone and asks the user to choose a device profile for use with Cisco Extension Mobility. Link: 

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmfeat/fsem.html 


Q138. Which option configures call preservation for H.323-based SRST mode? 

A. voice service voip h323 call preserve 

B. call preservation not possible with H.323 

C. call-manager-fallback preserve-call 

D. dial-peer voice 1 voip call preserve 

Answer:


Q139. Refer to the exhibit. 

Which CSS is used at the HQ Cisco Unified Communications Manager to reroute calls via the PSTN when the SAF network is unavailable? 

A. the phone device CSS 

B. the phone line CSS 

C. the phone line/device combined CSS 

D. the SAF CSS configured on the CCD requesting service 

E. the phone AAR CSS configured at the phone device 

F. No special CSS is required. If SAF patterns are accessible, the PSTN reroute is automatic. 

Answer:


Q140. What is the difference between an MGCP gateway and a SIP gateway? 

A. An MGCP gateway that dial peers be configured before PSTN calls can be placed and received. The SIP gateway requires no dial peers. 

B. An MGCP gateway can be added in Cisco Unified Communications Manager under the Gateway Type field using the gateway model. 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 MGCP gateway does not require a call agent for PSTN calls to be placed and received. 

D. An MGCP gateway can register with Cisco Unified Communications Manager. A SIP gateway will show status of "Unknown". 

E. The SIP gateway must be configured in Cisco Unified Communications Manager using a valid IP address on the gateway. The MGCP gateway must be configured in Cisco Unified 

Communications Manager using the domain name. 

Answer: