We provide real cisco 300 075 exam questions and answers braindumps in two formats. Download PDF & Practice Tests. Pass Cisco 300 075 dump Exam quickly & easily. The 300 075 pdf PDF type is available for reading and printing. You can print more and practice many times. With the help of our Cisco 300 075 vce dumps pdf and vce product and material, you can easily pass the 300 075 dump exam.
♥♥ 2021 NEW RECOMMEND ♥♥
Free VCE & PDF File for Cisco 300-075 Real Exam (Full Version!)
★ Pass on Your First TRY ★ 100% Money Back Guarantee ★ Realistic Practice Exam Questions
Free Instant Download NEW 300-075 Exam Dumps (PDF & VCE):
Available on:
http://www.surepassexam.com/300-075-exam-dumps.html
Q81. Which two locations are the best locations that an end user can use to determine if an IP phone is working in SRST mode? (Choose two.)
A. Cisco Unified Communications Manager Administration
B. IP phone display
C. Cisco Unified SRST Router
D. Cisco Unified MGCP Fallback Router
E. physical IP phone settings
Answer: B,E
Explanation:
Incorrect Answer: A, C, D IP Phone display and Physical phone IP settings are two locations were an end user can determine if an IP phone is working in SRST mode. Link: http://my.safaribooksonline.com/book/telephony/1587050757/survivable-remote-site-telephony-srst/529
Q82. 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
Q83. What is the difference between an MGCP gateway and a SIP gateway?
A. An MGCP gateway that dial peers be configured before PSTN calls can be placed and received. The SIP gateway requires no dial peers.
B. An MGCP gateway can be added in Cisco Unified Communications Manager under the Gateway Type field using the gateway model. The SIP gateway can connect to Cisco Unified Communications Manager only through a SIP trunk.
C. A SIP gateway requires a call agent for PSTN calls to be placed and received. An MGCP gateway does not require a call agent for PSTN calls to be placed and received.
D. An MGCP gateway can register with Cisco Unified Communications Manager. A SIP gateway will show status of "Unknown".
E. The SIP gateway must be configured in Cisco Unified Communications Manager using a valid IP address on the gateway. The MGCP gateway must be configured in Cisco Unified
Communications Manager using the domain name.
Answer: B
Q84. 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: A
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
Q85. 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
Q86. Refer to the exhibit
When the Cisco Unified Communications Manager advertises the Hosted DN Pattern, which pattern would be advertised?
A. 2XXX and the T0D1D will be 0:+498950555
B. 2XXX and the ToDID will be 0:+4989531 21
C. 4989S05552XXX and the ToDiD will be 0:
D. + 4989631 21 2XXX and the ToDiD will be 0:
E. Both +4989505552XXXand +4989531 21 2XXX will be advertised with ToDID of 0:
Answer: A
Explanation:
Incorrect Answer: B, C, D, E PSTN failover prepend digit is +498950555 Link:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_0_2/ccmfeat/fscallcontrol discovery.html
Q87. 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: A
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.
Q88. Which action configures PSTN backup for calls that are rejected by the gatekeeper CAC?
A. Configure AAR in Cisco Unified Communications Manager.
B. Configure CFUR in Cisco Unified Communications Manager.
C. Configure a route pattern, a route list, and route groups to a trunk and a gateway in Cisco Unified Communications Manager.
D. Configure a route pattern to a gateway in Cisco Unified Communications Manager.
Answer: C
Q89. Which command is needed to utilize local dial peers on an MGCP-controlled ISR during an SRST failover?
A. ccm-manager fallback-mgcp
B. telephony-service
C. dialplan-pattern
D. isdn overlap-receiving
E. voice-translation-rule
Answer: A
Q90. Which statement about enrollment in the IP telephony PKI is true? (Source. Understanding Cisco IP Telephony Authentication and Encryption Fundamentals)
A. CAPF enrollment supports the use of authentication strings.
B. The CAPF itself has to enroll with the Cisco CTL client.
C. LSCs are issued by the Cisco CTL client or by the CAPF.
D. MICs are issued by the CAPF itself or by an external CA.
Answer: A
Explanation:
Incorrect Answer: B, C, D
The CAPF enrollment process is as follows:
1. The IP phone generates its public and private key pairs.
2. The IP phone downloads the certicate of the CAPF and uses it to establish a TLS session with the CAPF.
3. The IP phone enrolls with the CAPF, sending its identity, its public key, and an optional authentication string.
4. The CAPF issues a certicate for the IP phone signed with its private key.
5. The CAPF sends the signed certicate to the IP phone.
Link: http://my.safaribooksonline.com/book/certification/cipt/9781587052613/understanding-cisco-ip-telephony-authentication-and-encryption-fundamentals/584.
