300-075 Exam Questions - Online Test


300-075 Premium VCE File

Learn More 100% Pass Guarantee - Dumps Verified - Instant Download
150 Lectures, 20 Hours

certleader.com

High quality of 300 075 dump answers materials and courses for Cisco certification for IT learners, Real Success Guaranteed with Updated 300 075 pdf pdf dumps vce Materials. 100% PASS CIPTV2 Implementing Cisco IP Telephony and Video, Part 2 exam Today!

P.S. High quality 300-075 courses are available on Google Drive, GET MORE: https://drive.google.com/open?id=1ZCQ4EawGfv2jUchm2y0Bw27VBlFAVB_S


New Cisco 300-075 Exam Dumps Collection (Question 5 - Question 14)

Q1. When implementing a dial plan for multisite deployments, what must be present for SRST to work successfully?

A. dial peers that address all sites in the multisite cluster

B. translation patterns that apply to the local PSTN for each gateway

C. incoming and outgoing COR lists

D. configuration of the gateway as an MGCP gateway

Answer: B


Q2. 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


Q3. 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: ACD

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.ht ml#wp1137460


Q4. When an incoming PSTN call arrives at an MGCP gateway, how does the called number get normalized to an internal directory number in Cisco Unified Communications Manager?

A. Normalization is done by configuring the significant digits for inbound calls on the MGCP gateway.

B. Normalization is done using route patterns.

C. Normalization is done using the gateway incoming called party prefixes based on number type.

D. Normalization is done using the gateway incoming calling party prefixes based on number type.

E. Normalization is achieved by local route group that is assigned to the MGCP gateway.

Answer: A


Q5. Which two actions are performed by the Call Control Discovery service after the local Cisco Unified Communications Manager loses its TCP connection with the primary and secondary Service Advertisement Framework? (Choose two.)

A. Calls are routed to the PSTN gateway after the Call Control Discovery Learned Pattern IP Reachable Duration parameter expires.

B. All learned patterns are purged from the local cache after the Call Control Discovery PSTN Failover Duration parameter expires.

C. The Service Advertisement Framework forwarder contacts all the remaining Service Advertisement Framework forwarders in the cluster.

D. All the remaining Service Advertisement Framework forwarders are notified for their learned patterns.

E. The Cisco Unified Communications Manager establishes a connection with the primary and secondary Service Advertisement Framework after the Learned Pattern IP Reachable Duration parameter expires.

F. Call Control Discovery immediately redirects all the calls to the PSTN gateway based on the learned patterns.

Answer: A,B


Q6. Which two entities could be represented by device mobility groups? (Choose two.)

A. countries

B. regions

C. directory numbers

D. transcoders

Answer: A,B


Q7. Which two configurations provide the best SIP trunk redundancy with Cisco Unified Communications Manager? (Choose two.)

A. Configure all SIP trunks with DNS SRV

B. Configure all SIP trunks with Cisco Unified Border Element

C. Configure all SIP trunks to point to a SIP gateway

D. Configure SIP trunks to be members of route groups and route lists

E. Configure all SIP trunks to allow TCP ports 5060

F. Configure all SIP trunks to point to a gatekeeper through SIP to H.323 gateway

Answer: A,D

Explanation: Incorrect: BCEF

For SIP trunks, Cisco Unified Communications Manager supports up to 16 IP addresses for each DNS SRV and up to 10 IP addresses for each DNS host name. The order of the IP addresses depends on the DNS response and may be identical in each DNS query. The OPTIONS request may go to a different set of remote destinations each time if a DNS SRV record (configured on the SIP trunk) resolves to more than 16 IP addresses, or if a host name (configured on the SIP trunk) resolves to more than 10 IP addresses. Thus, the status of a SIP trunk may change because of a change in the way a DNS query gets resolved, not because of any change in the status of any of the remote destinations.

Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a08sip.html


Q8. Which solution is needed to enable presence and extension mobility to branch office phones during a WAN failure?

A. SRST with MGCP fallback

B. SRST without MGCP fallback

C. Cisco Unified Communications Manager Express in SRST mode

D. SRST with VoIP dial peers to Cisco Unified Communications Manager Express

Answer: C


Q9. Which gateway does the Cisco Unified Communications Manager control all call activity?

A. SIP

B. MGCP

C. H.323

D. Media

Answer: B


Q10. An update of the configuration using the Cisco CTL client not needed when

.(SourcE. Configuring Cisco IP Telephony Authentication and Encryption)

A. a Cisco Unified CallManager has been removed

B. an LSC of the IP phone is upgraded

C. a security token is added to the system

D. an IP address of the Cisco TFTP server has been changed

Answer: B

Explanation: Incorrect: ACD

The CTL file contains entries for the following servers or security tokens:

u2022System Administrator Security Token (SAST)

u2022Cisco CallManager and Cisco TFTP services that are running on the same server

u2022Certificate Authority Proxy Function (CAPF)

u2022TFTP server(s)

u2022ASA firewall Link:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/security/8_6_1/secugd/secuauth.html#wp1028878


100% Most recent Cisco 300-075 Questions & Answers shared by Thedumpscentre, Get HERE: http://www.thedumpscentre.com/300-075-dumps/ (New 420 Q&As)