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Q121. Refer to the following exhibit. 

Which Cisco IOS SAF Forwarder configuration is correct? 

A. Exhibit A 

B. Exhibit B 

C. Exhibit C 

D. Exhibit D 

Answer:

Explanation: 

Incorrect Answer: B, C, D Summary steps to configure IOS SAF forwarder is given below 

1. enable 

2. configure terminal 

3. router eigrp virtual-instance-name 

4. service-family {ipv4 | ipv6} [vrf vrf-name] autonomous-system autonomous-system-number 

5. topology base 

6. external-client client_label 

7. exit-sf-topology 

8. exit-service-family

9. exit

10. service-family external-client listen {ipv4 | ipv6} tcp_port_number

11. external-client client-label

12. username user-name

13. password password-name 

14. 


Q122. Company A has deployed a VCS Control and is attempting to register a third-party endpoint. The engineer has confirmed that no traffic is being blocked for the endpoint and it is receiving a valid IP address. Which option could be the cause of this registration failure? 

A. Third-party endpoints are not compatible with VCS Control, only with VCS Expressway. 

B. Cisco Unified Communications Manager is required in addition to the VCS Control. 

C. An incorrect SIP domain is configured on the VCS Control for the endpoint. 

D. The VCS Control must be deployed together with VCS Expressway before endpoints can register to either one. 

Answer:


Q123. 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:


Q124. Which two configurations provide the best SIP trunk redundancy with Cisco Unified Communications Manager? (Choose two.) 

A. Configure all SIP trunks with DNS SRV 

B. Configure all SIP trunks with Cisco Unified Border Element 

C. Configure all SIP trunks to point to a SIP gateway 

D. Configure SIP trunks to be members of route groups and route lists 

E. Configure all SIP trunks to allow TCP ports 5060 

F. Configure all SIP trunks to point to a gatekeeper through SIP to H.323 gateway 

Answer: A,D 

Explanation: 

Incorrect Answer: B, C, E, F For SIP trunks, Cisco Unified Communications Manager supports up to 16 IP addresses for each DNS SRV and up to 10 IP addresses for each DNS host name. The order of the IP addresses depends on the DNS response and may be identical in each DNS query. The OPTIONS request may go to a different set of remote destinations each time if a DNS SRV record (configured on the SIP trunk) resolves to more than 16 IP addresses, or if a host name (configured on the SIP trunk) resolves to more than 10 IP addresses. Thus, the status of a SIP trunk may change because of a change in the way a DNS query gets resolved, not because of any change in the status of any of the remote destinations. 

Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a08sip.html 


Q125. 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 


Q126. Refer to the exhibit. 

The Cisco Unified Communications Manager at HQ has been configured for end-to-end RSVP. The Cisco Unified Communications Manager at BR 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 the BR site to the IP phone at the HQ site, which statement is true? 

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

D. The call will fail. 

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

Answer:


Q127. 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 


Q128. Refer to the exhibit. 

Which configuration elements must match for adjacent neighbors to establish a SAF neighbor relationship? 

A. the label name specified in the router eigrp command 

B. the autonomous-system number specified in the service-family ipv4 autonomous-system command 

C. the sf-interface configuration 

D. the topology base configurations 

E. the label name specified in the router eigrp command and the autonomous-system number 

Answer:

Explanation: 

Incorrect Answer: A, C, D, E service-family ipv4 autonomous-system 1 enables a Cisco SAF service family for the specified autonomous system on the router Link: 

http://www.cisco.com/en/US/docs/ios/saf/configuration/guide/saf_cg_ps10591_TSD_Products_Configuration_Guide_Chapter.html#wp1056363 


Q129. 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:

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 


Q130. If your IP telephony administrator asks you to configure a local zone for your dial plan to control the volume of calls between two end points in a centralized multisite environment, which two types of Call Admission Control can be implemented? (Choose two.) 

A. locations based 

B. automated alternate routing 

C. gatekeeper based 

D. SRST 

E. Cisco Unified Communications Manager based 

Answer: A,B 

Explanation: 

Incorrect Answer: C, D, E Location-based call admission control (CAC) manages WAN link bandwidth in Cisco Unified Communications Manager. Automated alternate routing (AAR) provides a mechanism to reroute calls through the PSTN or other network by using an alternate number. Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a02cac.html #wp1067747