Cisco 300-075 ExamCIPTV2 Implementing Cisco IP Telephony and Video, Part 2

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Q121. During device failover to the secondary Cisco Unified Communications Manager server, how does the phone recognize that the primary server is back? 

A. The secondary server keeps sending keepalive message to the primary server and when it succeeds, it unregisters the phones to force them to register to the primary. 

B. When the primary server goes online, it sends out an "ALIVE" message via broadcast so that the phones re-register. 

C. The phones never re-register with the primary server until the active (secondary) one goes down. 

D. The phone sends keepalive messages to the primary server frequently and when it succeeds, the phone re-registers with it. 

Answer:


Q122. When an H.323 trunk is added for Call Control Discovery, which statement is true? 

A. The H.323 trunk is added by selecting Inter-Cluster Trunk (Non-Gatekeeper Controlled) and Device Protocol Inter-Cluster Trunk. The Enable SAF check box should be selected in the trunk configuration. 

B. The H.323 trunk is added by selecting Inter-Cluster Trunk (Non-Gatekeeper Controlled) and Device Protocol Inter-Cluster Trunk. The Trunk Service Type should be Call Control Discovery. 

C. The H.323 trunk is added by selecting Call Control Discovery Trunk and then selecting 

H.323 as the protocol to be used. 

D. The H.323 trunk is added by selecting H.323 Trunk, and selecting Inter-Cluster Trunk as the Device Protocol. The destination IP address field is configured as ‘SAF’ to indicate that this trunk is used for SAF. 

Answer:

Explanation: 

Reference. Implementing Cisco Unified Communications Manager Part 2 (CIPT2), Chapter3: Implementing Multisite Connections, pg 70-71, Fig 3-14 and Fig 3-15 


Q123. Which statement about enrollment in the IP telephony PKI is true? (Source. Understanding Cisco IP Telephony Authentication and Encryption Fundamentals) 

A. CAPF enrollment supports the use of authentication strings. 

B. The CAPF itself has to enroll with the Cisco CTL client. 

C. LSCs are issued by the Cisco CTL client or by the CAPF. 

D. MICs are issued by the CAPF itself or by an external CA. 

Answer:

Explanation: 

Incorrect Answer: B, C, D 

The CAPF enrollment process is as follows: 

1. The IP phone generates its public and private key pairs. 

2. The IP phone downloads the certicate of the CAPF and uses it to establish a TLS session with the CAPF. 

3. The IP phone enrolls with the CAPF, sending its identity, its public key, and an optional authentication string. 

4. The CAPF issues a certicate for the IP phone signed with its private key. 

5. The CAPF sends the signed certicate to the IP phone. 

Link: http://my.safaribooksonline.com/book/certification/cipt/9781587052613/understanding-cisco-ip-telephony-authentication-and-encryption-fundamentals/584. 


Q124. Which three commands are mandatory to implement SRST for five Cisco IP Phones? (Choose three.) 

A. call-manager-fallback 

B. max-ephones 

C. keepalive 

D. limit-dn 

E. ip source-address 

Answer: A,B,E 


Q125. Which three options are overlapping parameters for roaming when a device is configured for Device Mobility? (Choose three.) 

A. MRGL 

B. location 

C. network locale 

D. codec 

E. extension 

F. device pool 

Answer: A,B,C 


Q126. Cisco Unified Communications Manager is configured with CAC for a maximum of 10 voice calls. 

Which action routes the 11th call through the PSTN? 

A. Configure an SIP trunk to the ISR. 

B. Configure Cisco Unified Communications Manager AAR. 

C. Configure Cisco Unified Communications Manager RSVP-enabled locations. 

D. Configure Cisco Unified Communications Manager locations. 

Answer:


Q127. Which solution is needed to enable presence and extension mobility to branch office phones during a WAN failure? 

A. SRST with MGCP fallback 

B. SRST without MGCP fallback 

C. Cisco Unified Communications Manager Express in SRST mode 

D. SRST with VoIP dial peers to Cisco Unified Communications Manager Express 

Answer:


Q128. 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 configuring the CFUR for the directory number that is applied to BR2 phone (+442288224001), the calls fail from the PSTN. Which two of the following configurations if applied to the router, would remedy this situation? (Choose two.) 

A. dial-peer voice 1 potsincoming called-number 228822…direct-inward-dialport 0/0/0:15 

B. dial-peer voice 1 potsincoming called-number 228822…direct-inward-dialport 0/0/0:13 

C. voice translation-rule 1rule 1 /228821 …$/ /+44&/exit!voice translation-profile pstn-intranslate called 1!voice-port 0/0/0:15translation-profile incoming pstn-in 

D. voice translation-rule 1rule 1 /228822…$/ /+44&/exit!voice translation-profile pstn-intranslate called 1!voice-port 0/0/0:15translation-profile incoming pstn-in 

E. The router does not need to be configured 

Answer: A,D 


Q129. Refer to the exhibit. 

The HQ site uses area code 650. The BR1 site uses area code 408. The long distance national code for PSTN dialing is 1. To make a long distance national call, an HQ or BR1 user dials access code 9, followed by 1, and then the 10-digit number. 

Both sites use MGCP gateways. AAR must use globalized call routing using a single route pattern. Assume that all outgoing PSTN numbers are localized at the egress gateway as shown in the exhibit. 

How many route lists and route groups should be configured for AAR at a minimum? 

A. a single route list with a local route group for each site 

B. two route lists and two route groups for each site 

C. a single route list and four route groups for each site 

D. None. The AAR CSS can point directly to the route pattern. 

Answer:


Q130. 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:


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