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New Cisco 300-075 Exam Dumps Collection (Question 4 - Question 13)
Question No: 4
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
Question No: 5
Which statement about the function of a gatekeeper is true?
A. A gatekeeper improves call routing between servers within a single Cisco Unified Communications Manager cluster.
B. A gatekeeper can replace the dial plan of a Cisco Unified Communications Manager cluster.
C. A gatekeeper can simplify the dial plan between many different Cisco Unified Communications Manager clusters.
D. Gatekeepers can be implemented to deploy RSVP-based CAC.
Answer: C
Question No: 6
Which technologies provide remote-site redundancy for Cisco IP Phones during a WAN failure?
A. SRST and MGCP fallback
B. SRST and TEHO
C. TEHO and MGCP fallback
D. SRST and AAR
Answer: A
Question No: 7
Which two statements are true regarding the implementation of globalized call-routing in terms of localized call egress? (Choose two.)
A. Calling-party numbers are routed from the gateway or trunks to phones.
B. Called-party numbers are routed from the gateway or trunks to phones.
C. Calling-party numbers of internal calls are routed from the gateway or trunks.
D. Calling-party calls are routed to the gateway and trunks.
Answer: A,D
Question No: 8
Which two have to be defined in the Forward All field? (Choose two.) (SourcE. Preventing Toll Fraud)
A. calling search space
B. destination
C. partition
D. hunt list
Answer: A,B
Explanation: Incorrect: CD
Destinationu2014This setting indicates the directory number to which all calls are forwarded. Use any dialable phone number, including an outside destination.
Calling Search Spaceu2014This setting applies to all devices that are using this directory number.
Question No: 9
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
Question No: 10
Which statement about Service Advertisement Framework is true?
A. SAF requires that the EIGRP be configured on all routers, including non-SAF routers.
B. SAF requires that the EIGRP be configured only on SAF routers. Non-SAF routers act as an IP cloud.
C. SAF has no dependency on the underlying routing protocol, as long as it is a dynamic routing protocol. Static routes are not supported.
D. SAF operates on any dynamic or static IP routing configuration. SAF is totally independent of the underlying routing protocol.
Answer: D
Explanation:
Because Cisco SAF is independent of IP routing and uses underlying Cisco routing technology to distribute service advertisements in a reliable and efficient manner, Cisco SAF will run in networks over any routing protocol they may have in place such as Enhanced Interior Gateway Routing Protocol (EIGRP), Open Shortest Path First (OSPF), Exterior Border Gateway Protocol (EBGP) over an MPLS service, or static routing (Figure 2).
http://www.cisco.com/en/US/prod/collateral/iosswrel/ps6537/ps6554/ps6599/ps10822/whitepaper_c11-636604.html
Question No: 11
Refer to the exhibit.
How many calls are permitted by the RSVP configuration?
A. one G.711 call
B. two G.729 calls
C. one G.729 call and one G.711 call
D. eight G.729 calls
E. four G.729 calls
Answer: B
Explanation: Incorrect: ACDE
In performing location bandwidth calculations for purposes of call admission control, Cisco Unified Communications Manager assumes that each call stream consumes the following amount of bandwidth:
u2022G.711 call uses 80 kb/s.
u2022G.722 call uses 80 kb/s.
u2022G.723 call uses 24 kb/s.
u2022G.728 call uses 26.66 kb/s.
u2022G.729 call uses 24 kb/s.
Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a02cac.html
#wpxref28640
Question No: 12
Refer to the exhibit.
How does the Cisco Unified Communications Manager advertise dn-block 1?
A. 4XXX and the ToDID will 0:
B. 4XXX and the ToDID will 0:1972555
C. 4XXX
D. 4XXX and the ToDID will 0:+ 1972555
E. 19725554XXX
Answer: B
Question No: 13
You are deploying a remote office setup that connects with Cisco Unity Communications Manager at a hub location. You have an available dedicated bandwidth of 20% from the 2- Mb/s WAN circuit for VoIP that supports a maximum of 17 calls. Which codec do you configure in Cisco Unity Communications Manager to achieve this?
A. G.722
B. G.711
C. G.729
D. iSAC
E. GSM-FR
F. iLBC
Answer: C
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