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Q1. 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 

Q2. Refer to the exhibit. 

IT shows an H.323 gateway configuration in a Cisco Unified Communications Manager environment. An inbound PSTN call to this H.323 gateway fails to connect to IP phone extension 2001. The PSTN user hears a reorder tone. Debug isdn q931 on the H.323 gateway shows the correct called-party number as 5015552001. Which two configuration changes can correct this issue? (Choose two.) 

A. Add port 1/0:23 to dial-peer voice 123 pots. 

B. Ensure that the Significant Digits for inbound calls on the H.323 gateway configuration is 4. 

C. Add a voice translation profile to truncate the number from 10 digits to 4 digits. Apply the voice translation profile to the Voice-port. The configuration field "Significant Digits for inbound calls" is left at default (All). 

D. Add the command h323-gateway voip id on interface vlan120. 

E. Change the destination-pattern on the dial-peer voice 23000 VoIP to 501501? and change the Significant Digits for inbound calls to 4. 

Answer: B,E 

Explanation: 

Incorrect Answer: A, C, D Choose the number of significant digits to collect, from 0 to 32. Cisco Unified Communications Manager counts significant digits from the right (last digit) of the number that is called. Link: http://cisco.biz/en/US/docs/voice_ip_comm/cucmbe/admin/8_6_1/ccmcfg/b06trunk.html 

Q3. Refer to the exhibit. 

Assuming the regions configuration to BR only permits G.729 codec, how many calls are allowed for the BR location? 

A. Total of four calls; two incoming and two outgoing. 

B. Total of two calls in either direction. 

C. Total of four calls to the BR location. Outgoing calls are not impacted by the location configuration. 

D. Total of four calls in either direction. 

E. Two outgoing calls. Incoming calls are unlimited. 

Answer: D 

Explanation: 

Incorrect Answer: A, B, C, E In performing location bandwidth calculations for purposes of call admission control, Cisco Unified Communications Manager assumes that each G.729 call stream consumes 24 kb/s amount of bandwidth Link: 

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

Q4. When an incoming PSTN call arrives at an H.323 gateway, how does the called number get normalized to an internal directory number in Cisco Unified Communications Manager? 

A. Normalization is done by configuring the significant digits for inbound calls on the H.323 gateway configuration in Cisco Unified Communications Manager. 

B. Normalization is done using route patterns. 

C. Normalization is done using the gateway incoming calling party prefixes based on number type. 

D. Normalization is achieved by local route group that is assigned to the H.323 gateway. 

Answer: A 

Q5. When an incoming PSTN call arrives at an H.323 gateway, how does the calling number get normalized to a global E.164 number with + prefix in Cisco Unified Communications Manager? 

A. Normalization is done using translation patterns. 

B. Normalization is done using route patterns. 

C. Normalization is done using the gateway incoming called party prefixes based on number type. 

D. Normalization is done using the gateway incoming calling party prefixes based on number type. 

E. Normalization is achieved by local route group that is assigned to the H.323 gateway. 

Answer: D 

Q6. Refer to the exhibit. 

What is the correct value to use for the "DSCP for TelePresence Calls" Cisco CallManager service parameter? 

A. 28 (011100) 

B. 34 (100010) 

C. 41 (101001) 

D. 46 (101110) 

Answer: A 

Q7. Which two options for a Device Mobility-enabled IP phone are true? (Choose two.) 

A. The phone configuration is not modified. 

B. The roaming-sensitive parameters of the current (that is, the roaming) device pool are applied. 

C. The user-specific settings determine which location-specific settings are downloaded from the Cisco Unified Communications Manager device pool. 

D. If the DMGs are the same, the Device Mobility-related settings are also applied. 

Answer: B,D 

Q8. Which statement is true when considering a Cisco VoIP environment for regional configuration? 

A. G.711 requires 128K of bandwidth per call. 

B. G.729 requires 24K of bandwidth per call. 

C. The default codec does not matter if you have defined a hardware MTP in your Cisco Unified Communications Manager environment. 

D. To deploy a Cisco H.323 gatekeeper, you must configure MTP resources on the gatekeeper and only use G.711 between regions. 

Answer: C 

Q9. 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 

What two issues could be causing the Cisco Jabber Video for TelePresence failure shown in the exhibit? (Choose two.) 

A. Incorrect username and password 

B. Wrong SIP domain configured. 

C. User is not associated with the device. 

D. IP or DNS name resolution issue. 

E. CSF Device is not registered. 

F. IP Phone DN not associated with the user. 

Answer: B,D 

Q10. Refer to the exhibit: 

The exhibit shows a SAF Forwarder configuration attached to a Cisco Unified Communications Manager. 

Which minimum configuration for a Cisco Unified Communications Manager Express SAF Forwarder is needed to establish a SAF neighbor relationship with this SAF Forwarder? 

A. router eigrp SAFiservice-family ipv4 autonomous-system 1!topology baseexit-sf-topologyexit-service-familyvoice service safprofile trunkroute 1session protocol sip interface Loopback1 transport tcp port 5060! 

B. router eigrp SAF!service-family ipv4 autonomous-system 1!topology baseexit-sf-topologyexit- service-family!voice service safprofile trunk-route 1session protocol sip interface Loopback1 transport tcp port 5060!profile dn-block 1 alias-prefix 1972555pattern 1 type extension 4xxx!profile callcontrol 1dn-servicetrunk-route 1dn-block 1dn-block 2!channel 1 vrouter SAF asystem 1subscribe callcontrol wildcardedpublish callcontrol 1! 

C. router eigrp SAF!service-family ipv4 autonomous-system 1!topology baseexit-sf-topologyexit-service-family! 

D. None of above configurations contain sufficient information. 

Answer: C 

Explanation: 

Incorrect Answer: A, B, D only following configuration is enough router eigrp SAF service-family ipv4 autonomous-system 1 exit-service-family link: 

http://www.cisco.com/en/US/prod/collateral/iosswrel/ps6537/ps6554/ps6599/ps10822/whitepaper_c11-636604.html