Master the cisco 300 075 Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2) content and be ready for exam day success quickly with this Ucertify 300 075 vce exam fees. We guarantee it!We make it a reality and give you real 300 075 pdf questions in our Cisco 300 075 dump braindumps.Latest 100% VALID Cisco 300 075 ciptv2 Exam Questions Dumps at below page. You can use our Cisco 300 075 dump braindumps and pass your exam.


2026 New 300-075 Exam Dumps with PDF and VCE Free: https://www.2passeasy.com/dumps/300-075/

Q1. Video calls using 384 kbps need to be supported across a gatekeeper-controlled trunk. What value should be entered into the gatekeeper to support this bandwidth? 

A. 768 kbps 

B. 384 kbps 

C. 512 kbps 

D. 192 kbps 

Answer:

Explanation: 

Incorrect Answer: A, C, D A 384-kb/s video call may comprise G.711 at 64 kb/s (for audio) plus 320 kb/s (for video). This sum does not include overhead. If the audio codec for a video call is G.729 (at 24 kb/s), the video rate increases to maintain a total bandwidth of 384 kb/s. Link: 

http://www.ciscosystems.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a08video.html#wp1059726 

Q2. When Cisco Extension Mobility is implemented, how is the audio source for the MOH selected? 

A. The audio source that is configured at the user device profile is selected. 

B. The audio source that is configured at the home phone of the user is selected. 

C. The audio source that is configured at the physical phone used for the Cisco Extension Mobility login is selected. 

D. The audio source that is configured in the IP Voice Media Streaming parameters is selected. 

Answer:

Explanation: 

Incorrect Answer: B, C, D To specify the audio source that plays when a user initiates a hold action, choose an audio source from the User Hold MOH Audio Source drop-down list box from device profile configuration settings. Link: http://cisco.biz/en/US/docs/voice_ip_comm/cucmbe/admin/8_6_1/ccmcfg/b06dvprf.html 

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. 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 

Q5. 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. 

Which partition should be configured in the AAR CSS applied at the phones? 

A. PSTN partition 

B. LD partition 

C. The HQ AAR CSS must include a partition assigned to route pattern 91408XXXXXXX. The BR1 AAR CSS must include a partition assigned to route pattern 91650XXXXXXX. 

D. AAR CSS must contain translation pattern 9.1[2-9]XX[2-9]XXXXXX for each site that must be globalized. Otherwise the called numbers will not be localized at the egress gateway. 

Answer:

Q6. While configuring Call Survivability in Cisco Unified Communications Manager, what step is mandatory to reach remote sites while in SRST mode? 

A. Enable Cisco Remote Site Reachability. 

B. Configure CFUR. 

C. Enable the SRST checkbox in the MGCP gateway. 

D. Configure the H.323 gateway for SRST in Cisco Unified Communications Manager. 

E. Enable the Failover Service parameter. 

Answer:

Explanation: 

Incorrect Answer: A, C, D, E Call Forward Unregistered (CFUR) functionality provides the automated rerouting of calls through the PSTN when an endpoint is considered unregistered due to a remote WAN link failure Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/models.html 

Q7. 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 

Q8. Which two options for a Device Mobility-enabled IP phone are true? (Choose two.) 

A. The phone configuration is not modified. 

B. The roaming-sensitive parameters of the current (that is, the roaming) device pool are applied. 

C. The user-specific settings determine which location-specific settings are downloaded from the Cisco Unified Communications Manager device pool. 

D. If the DMGs are the same, the Device Mobility-related settings are also applied. 

Answer: B,D 

Q9. Which system configuration is used to set a restriction on audio bandwidth? 

A. region 

B. location 

C. physical location 

D. licensing 

Answer:

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