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Q1. Which three of the following are steps in configuring MGCP Fallback and Cisco Unified SRST? (Choose three)
A. Define the SRST reference for phones in the Device Pool configuration
B. Enable and configure the MGCP fallback and Cisco Unified SRST features on the IOS gateways.
C. Implement a simplified SRST dial plan on the remote-site-gateways to ensure connectivity for remote-site phones in SRST mode.
D. Enable SIP trunking between both remote and hub sites to provide mesh coverage.
E. Define the SRST reference in the configuration on the IP Phones.
F. Enable and configure the MGCP fallback on the IOS gateway but not Cisco Unified SRST since it is enabled automatically.
Answer: A,B,C
Q2. Which two actions ensure that the call load from Cisco TelePresence Video Communication Server to a Cisco Unified Communications Manager cluster is shared across Unified CM nodes? (Choose two.)
A. Create a neighbor zone in VCS with the Unified CM nodes listed as location peer addresses.
B. Create a single traversal client zone in VCS with the Unified CM nodes listed as location peer addresses.
C. Create one neighbor zone in VCS for each Unified CM node.
D. Create a VCS DNS zone and configure one DNS SRV record per Unified CM node.
E. In VCS set Unified Communications mode to Mobile and remote access and configure each Unified CM node.
Answer: A,D
Q3. 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
Q4. 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.
Q5. Which statement best describes globalized call routing in Cisco Unified Communications Manager?
A. All incoming calling numbers on the phones are displayed as an E.164 with the + prefix.
B. Call routing is based on numbers represented as an E.164 with the + prefix format.
C. All called numbers sent out to the PSTN are in E.164 with the + prefix format.
D. The CSS of all phones contain partitions assigned to route patterns that are in global format.
E. All phone directory numbers are configured as an E.164 with the + prefix.
Answer: B
Explanation:
Incorrect Answer: A, C, D, E For the destination to be represented in a global form common to all cases, we must adopt a global form of the destination number from which all local forms can be derived. The + sign is the mechanism used by the ITU's E.164 recommendation to represent any PSTN number in a global, unique way. This form is sometimes referred to as a fully qualified PSTN number. Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/dialplan.html#wp1153205
Q6. Which action configures transcoding resources in Cisco Unified Communications Manager to function with branch office Cisco IP Phones?
A. Configure the branch office IP phones with CSS and partitions.
B. Configure the branch office IP phones with MRGs and MRGLs.
C. Configure the branch office IP phones with regions.
D. Configure the branch office IP phones with locations.
Answer: B
Q7. 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
Q8. What are the two tasks that you must perform to configure the Service Advertisement Framework forwarder in Cisco Unified Communications Manager? (Choose two.)
A. create VPN groups
B. create VPN profiles
C. create a new Service Advertisement Framework security profile
D. set feature configuration parameters of Call Control Discovery
E. configure Service Advertisement Framework forwarder information
F. enable enterprise parameter for Service Advertisement Framework forwarder
Answer: C,E
Q9. What is the difference between an H.323 gateway and a SIP gateway?
A. An H.323 gateway requires that dial peers be configured before PSTN calls can be placed and received. The SIP gateway requires no dial peers.
B. The H.323 gateway can be added in Cisco Unified Communications Manager under gateway type as H.323 Gateway. 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 H.323 gateway does not require a call agent for PSTN calls to be placed and received.
D. An H.323 gateway can register with Cisco Unified Communications Manager. A SIP gateway will show status of "Unknown".
E. The H.323 gateway must be configured in Cisco Unified Communications Manager using a valid IP address on the gateway. The SIP gateway must be configured in Cisco Unified Communications Manager using the domain name.
Answer: B
Q10. Which sign is prefixed to the number in global call routing?
A. -
B. +
C. #
D. @
E. &
F. *
Answer: B
Q11. 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 is controlling the SX20, the Cisco TelePresence MCU, and the Cisco Jabber TelePresence for Windows.
Use the exhibits to answer the following questions.
SX20 System Information
DX650 Configuration
MRGL
DP
Locations
AARG
CSS
Movi Failure
Movi Settings
A new DX650 IP phone with MAC address D0C7.8914.132D, IP address is 172.18.32.119 has been added to the Cisco Unified Communications Manager, but is not registering properly.
What is causing this failure?
A. Device Pool cannot be default.
B. The DX650 is the incorrect calling search space.
C. The DX650 Phones does not support SIP.
D. The location Hub_None has not been activated.
E. The DX650’s MAC address is incorrect in the Cisco UCM.
Answer: E
Explanation:
From the screen capture below we can see that the DX650 phone’s MAC address was incorrectly entered as D0C78914131D, not D0C78914132D.
Q12. In a node-specific Service Advertisement Framework forwarder deployment model, what is the maximum number of Service Advertisement Framework forwarders that you can assign to a specific node?
A. 1
B. 2
C. 3
D. 4
E. 5
F. 6
Answer: B
Q13. 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
Q14. When a SIP trunk is added for Call Control Discovery, which statement is true?
A. The SIP trunk is added by selecting SIP Trunk and SIP Protocol. The Enable SAF check box should be selected.
B. The SIP trunk is added by selecting SIP Trunk and SIP Protocol. The Trunk Service Type should be Call Control Discovery.
C. The SIP trunk is added by selecting Call Control Discovery Trunk and then selecting SIP as the protocol to be used.
D. The SIP trunk is added by selecting SIP Trunk and SIP Protocol. The destination IP address field is configured as ‘SAF’ to indicate that this trunk is used for SAF.
Answer: B
Q15. 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 adding SRST functionality the SRST does not work. After reviewing the exhibits, which of the following reasons could be causing this failure?
A. Device Pool cannot be default.
B. The Cisco UCM is pointing to the wrong IPv4 address of the BR router.
C. The router does not support SRST.
D. The SRST enabled router is not configured correctly.
Answer: A