300-075 Exam - Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2)

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Q1. 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 

A third collaboration call fails between the backbone site and the HQ site. After reviewing the exhibits, which of the following reasons could be causing this failure? 

A. Not enough bandwidth has been allocated. 

B. Device Pool. 

C. Location. 

D. The pipe is not functioning. 

Answer:

Explanation: 

Based on the exhibit, each call is limited to no more than 128kbps per call, but the total available bandwidth is set to 256 kbps. This will allow the first to calls to go through, but there will be no more available bandwidth for the third call. 

Q2. Refer to the following exhibit. 

The MGCP gateway has the following configurations: 

called party transformation CSS HQ_cld_pty CSS (partition=HQ cld_pty.Pt) call.ng party transformation CSS HQ_clng_pty CSS (partition=HQ_clng_pty Pt) 

All translation patterns have the check box "Use Calling Party's External Phone Number Mask" enabled. 

When the IP phone at extension 3001 places a call to 9011 49403021 56001# what is the resulting called and calling number that is sent to the PSTN? 

A. The called number is 01 1 49403021 56001. The calling number will be 5553001 and number type set to subscriber. 

B. The called number is 011 49403021 56001. The calling number will be 5215553001 and number type set to national. 

C. The called number is 4940302156001 with number type set to international. The calling number will be 5215553001 and number type set to national. 

D. The called number is +49403021 56001 with number type set to international. The calling number will be 5215553001 and number type set to subscriber. 

Answer:

Explanation: 

Incorrect Answer: B, C, D Check the check box "Use Calling Party's External Phone Number Mask" if you want the full, external phone number to be used for calling line identification (CLID) on outgoing calls. You may also configure an External Phone Number Mask on all phone devices. Link: http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a00805 b6f33.shtml 

Q3. 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 

Which device configuration option will allow an administrator to control bandwidth between calls placed between branches? 

A. Media Resource Group List 

B. Device Pool 

C. Location 

D. AAR Group 

E. Regions 

Answer:

Explanation: 

In Cisco Unified Communications Manager Administration, use the System > Location Info menu path to configure locations. Use locations to implement call admission control in a centralized call-processing system. Call admission control enables you to regulate audio quality and video availability by limiting the amount of bandwidth that is available for audio and video calls over links between the locations 

Q4. When considering Extension Mobility, what happens if a user logs into a phone for which the user does not have a user device profile? 

A. The phone reboots with an error. 

B. If a default device profile for this phone has been configured, it is loaded. 

C. The user cannot log in. 

D. Another user device profile is loaded. 

Answer:

Q5. Which two statements about SAF service identifier numbers are true? (Choose two.) 

A. They are generated in the format service:sub-service:instance.instance.instance.instance. 

B. They are 16-bit decimal identifiers. 

C. They are generated in the format data-source:sub-service:instance.matrix.fifty.saf. 

D. They are 32-bit decimal identifiers. 

E. They are generated in the format data.saf.cucm-publisher.asf@domain.local. 

F. They are generated in the format telco.cisco.saf-forwader.db.replicate.data.local. 

Answer: A,B 

Q6. Refer to the exhibit. 

The exhibit shows centralized Cisco Unified Communications Manager configuration components for TEHO calls to U.S. area code 408 from the U.K. The PSTN access code for the U.K. is 9 and 001 for international calls to the U.S. What should the TEHO-US route list configuration consist of? 

A. First route group should point only to the U.K. gateway. The second route group should point to the U.S. gateway. 

B. First route group should be only the local route group. The second route group should point to the U.S. gateway. 

C. First route group should point only to the U.S. gateway. The second route group should be the local route group. 

D. The TEHO-US route list should contain only the local route group. The globalized configuration means that the appropriate gateway will be selected automatically. 

E. The \+! route pattern should point directly to the U.S. gateway. 

Answer:

Explanation: 

Incorrect Answer: A, B, D The route group points to one or more gateways and can choose the gateways for call routing based on preference. The route group can serve as a trunk group by directing all calls to the primary device and then using the secondary devices when the primary is unavailable. One or more route lists can point to the same route group. Link: 

http://www.ciscosystems.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a08 gw.html#wp1167274 

Q7. Which two options are requirements for deploying an H.323 gateway with Cisco Unified Communications Manager? (Choose two.) 

A. Cisco Unified Communications Manager and the H.323 gateway must be configured use the same TCP port for H.323 calls. 

B. The H.245TCSTimeout timer must be set to at least 25. 

C. Cisco voicemail ports must be active. 

D. The Media Exchange Interface Capability Timer must be set to less than 20. 

E. The Media Exchange Timer must be set to less than 20. 

Answer: A,B 

Q8. When you configure QoS on VCS, which settings do you apply if traffic through the VCS should be tagged with DSCP AF41? 

A. Set QoS mode to DiffServ and tag value 32. 

B. Set QoS mode to IntServ and tag value to 34. 

C. Set QoS mode to DiffServ and tag value 34. 

D. Set QoS mode to IntServ and tag value to 32. 

E. Set QoS mode to ToS and tag value to 32. 

Answer:

Q9. Which three tests can you perform to verify redundancy in the customer environment? (Choose three.) 

A. Verify that all phones are registered to a second subscriber server. 

B. Verify that media resources fail over to a secondary subscriber server when the publisher fails. 

C. Verify that Cisco Unified IP phones running SCCP go into SRST mode when the WAN connection is disconnected. 

D. Verify that HSRP is active on the Cisco Unified Communications Manager subscriber servers. 

E. Verify that the H.323 redundant connection is active. 

F. Verify that SCCP fallback is configured in Cisco Unified Communications Manager. 

Answer: A,B,C 

Q10. Assume that local route groups are configured. When an IP phone moves from one device mobility group to another, which two configuration components are not changed? (Choose two.) 

A. IP subnet 

B. user settings 

C. SRST reference 

D. region 

E. phone button settings 

Answer: B,E 

Explanation: 

Incorrect Answer: A, C, D Although the phone may have moved from one subnet to another, the physical location and associated services have not changed. Link: 

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_1_2/ccmfeat/fsdevmob.html#wp1137460