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Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2) Certification Exam

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Cisco 300-075 Free Practice Questions

Q1. 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:

Q2. What is the purpose of configuring a hardware-based MTP when deploying Cisco Unified Communications Manager? 

A. to allow for supplementary services such as hold, transfer, and conferencing 

B. when you need support for up to 24 MTP sessions on the same server and 48 on a separate server 

C. when you need the ability to grow support by using DSPs 

D. when you want to only use Cisco Unified Communications Manager resources 

Answer:

Q3. 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:

Q4. Which bandwidth amounts are correct for configuring locations? 

A. 8 kb/s for G.729, 64 kb/s for G.711, and 64 kb/s for G.722 

B. 8 kb/s for G.729, 64 kb/s for G.711, and 16 kb/s for G.722 

C. 64 kb/s for G.729, 64 kb/s for G.711, and 64 kb/s for G.722 

D. 8 kb/s for G.729, 8 kb/s for G.711, and 8 kb/s for G.722 

Answer:

Q5. Cisco Unified border element is configured to support RSVP-based CAC. When is the RSVP path and reservation message sent and received? 

A. Immediately after the call setup message is received and the reservation message is received after H.245 capabilities negotiation is completed. 

B. The path and reservation messages are sent and received after the H.245 capabilities negotiation is completed. 

C. The path and reservation messages are sent and received immediately after the call setup message is received. 

D. The path is setup once the global command call rsvp-sync is configured. 

Answer:

Q6. 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:

Q7. 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 

Q8. When using Cisco Unified Communications Manager Express in SRST mode, how many multicast music on hold streams can be utilized by the system at any given time? 

A. 3 

B. 6 

C. 2 

D. 4 

E. 1 

F. 5 

Answer:

Q9. Refer to the exhibit. 

What should the destination IP address be configured as on the HQ and BR1 SIP trunks? 

A. The HQ SIP trunk destination IP address should be 10.1.6.10. The BR1 SIP trunk destination IP address should be 10.1.5.10. 

B. The destination IP address is not configurable. Each cluster will learn about the remote trunk IP address through SAF learned routes. 

C. The destination IP address will be learned automatically and configured on the SIP trunks after the Cisco Unified Communications Managers discover themselves. 

D. The HQ SIP trunk destination IP address should be the HQ SAF Forwarder IP address. The BR1 SIP trunk destination IP address should be the BR1 SAF Forwarder IP address. 

Answer:

Explanation: 

Incorrect Answer: A, C, D 

The gatekeeper changes the IP address of this remote device dynamically to reflect the IP address of the remote device. 

Link: 

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a08trnk.html 

Q10. How is the region assigned to a device such as an IP phone? 

A. Regions are assigned directly in the device configuration page. 

B. Regions can be assigned only through a device pool. 

C. Regions can be assigned either directly on the device configuration page or through the device pool. If both configurations exist, the device pool region configuration takes precedence. 

D. Regions can be assigned either directly on the device configuration page or through the device pool. If both configurations exist, the device region configuration takes precedence. 

Answer:

Q11. Refer to the exhibit. 

When a call between two HQ users is being conferenced with a remote user at BR, which configuration is needed? 

A. The BR_MRG must contain the transcoder device. The BR_MRGL must be assigned to the BR phones. 

B. The HQ_MRG must contain the transcoder device. The HQ_MRGL must be assigned to the HQ phones. 

C. A transcoder should be configured at the remote site and assigned to all remote phones through the BR_MRGL. 

D. The HQ_MRG must contain the transcoder device. The HQ_MRGL must be assigned to the software conference bridge. 

E. Enable the software conference bridge to support the G.711 and G.729 codecs in Cisco Unified Communications Manager Service Parameters. 

Answer:

Q12. Which two are gatekeeper-controlled trunk options that support gatekeeper call administration control? (Choose two.) 

A. H.323 

B. H.245 

C. H.225 

D. intercluster 

E. intracluster 

Answer: C,D 

Q13. Which three steps configure Cisco Unified Survivable Remote Site Telephony for SIP phones? (Choose three.) 

A. Configure a phone NTP reference. 

B. Configure an SRST reference. 

C. Configure the SIP registrar. 

D. Configure voice register global dn. 

E. Configure voice register pool. 

F. Configure telephony service. 

Answer: B,C,E 

Q14. 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 

Q15. 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 

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