Exam Code: 300-075 (Practice Exam Latest Test Questions VCE PDF)
Exam Name: Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2)
Certification Provider: Cisco
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Q31. Which sign is prefixed to the number in global call routing? 

A. -

B. + 

C. # 

D. @ 

E. & 

F. * 

Answer: B 


Q32. When you configure Cisco Unified Communications Manager, you need to configure the router for Survivable Remote Site Telephony in case the Cisco Unified Communications Manger stops working. On which two factors would the number of IP phones and Directory Numbers that can register to the SRST router depend? (Choose two.) 

A. The protocol that is used in Cisco Unified Communications Manager 

B. Cisco Unified Communications Manager version 

C. Cisco IOS Software version 

D. WAN link bandwidth 

E. capacity of the Cisco Media Convergence Server 

F. router platform 

Answer: C,F 


Q33. 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 


Q34. How do RSVP-enabled locations differ from Cisco Unified Communications Manager locations? 

A. RSVP is configured in the ISR independent of Cisco Unified Communications Manager. 

B. RSVP enables AAR within Cisco Unified Communications Manager. 

C. RSVP is topology aware. 

D. RSVP is configured in Cisco Unified Communications Manager independent of the ISR. 

Answer: C 


Q35. 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 


Rebirth 300-075 test question:

Q36. When an external call is placed from Ajax, they would like the ANI that is sent to the PSTN to be the main number, not the extension. For domestic calls, they would like 10 digits sent; for international calls, they would like to send the country code 1 and the 10 digits. How can this be accomplished? 

A. Add a translation pattern to the dial peers in the gateway that adds the appropriate digits to the outgoing ANI. 

B. In the external call route patterns, set the external phone number mask to the main number. Use 10 digits in the domestic route pattern and 1 followed by the main number digits in the international route patterns. 

C. Use a calling party transform mask for each route group in the corresponding route list configuration. Set the explicit 10-digit main number for domestic calls and 1 followed by the main number for the international route patterns. 

D. In the directory number configurations, set the prefix digits field to the country code and the 10 digits of the main number. This will be truncated to the 10-digit number for domestic calls and sent out in its entirety for international calls. 

Answer: C 

Explanation: 

Incorrect Answer: A, B, D calling party transformation mask value is Valid entries for the NANP include the digits 0 through 9; the wildcard characters X, asterisk (*), and octothorpe (#); and the international escape character +. Link: 

http://www.cisco.com/en/US/docs/voice_ip_comm/cucmbe/admin/8_6_1/ccmcfg/b03trpat.ht ml 


Q37. 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 7965 and 9971 Video IP Phones. The Cisco VCS and TMS control the Cisco TelePresence Conductor, the Cisco TelePresence MCU, and the Cisco Jabber TelePresence for Windows. 

Use the exhibits to answer the following questions. 

DNS Servers 

Device Pool 

Expressway 

ILS 

Locations 

MRA 

Speed Dial 

SIP Trunk 

Which three configuration tasks need to be completed on the router to support the registration of Cisco Jabber clients? (Choose three.) 

A. The DNS server has the wrong IP address. 

B. The internal DNS Service (SRV) records need to be updated on the DNS Server. 

C. Flush the DNS Cache on the client. 

D. The DNS AOR records are wrong. 

E. Add the appropriate DNS SRV for the Internet entries on the DNS Server. 

Answer: B,C,E 


Q38. Refer to the exhibit. 

When the user of a phone registered to the Cisco Unified Communications Manager places a call to 3001 when the SAF network is down, what happens? 

A. The call fails. 

B. The call is rerouted to the PSTN with the constructed PSTN number as +442288223001 

C. The call is rerouted to the PSTN with the constructed PSTN number as 442288223001 

D. The call is rerouted to the PSTN with the constructed PSTN number as 0002288223001 

E. The call is rerouted to the PSTN with the constructed PSTN number as +0002288223001 

Answer: A 

Explanation: 

Incorrect Answer: B, C, D When the SAF forwarder loses network connection with its call-control entity, the SAF forwarder withdraws those learned patterns that were published by the call control entity. In this case, CCD requesting service marks those learned patterns as unreachable via IP, and the calls get routed through the PSTN gateway. Link: 

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmfeat/fscallcontrol discovery.html 


Q39. 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: C 


Q40. Which statement is true regarding the configuration of SAF Forwarder? 

A. In a multisite dial plan, SAF Forwarders may exist in multiple autonomous systems. 

B. The client label that is configured in Cisco Unified Communications Manager must match the configuration on the SAF Forwarder router. 

C. There should not be multiple nodes of Cisco Unified Communications Manager clusters acting as SAF clients. 

D. The destination IP address must match the loopback address of the SAF router. 

Answer: A