Exam Code: 642-427
Exam Name: Troubleshooting Cisco Unified Communications v8.0 (TVOICE v8.0)
Certification Provider: Cisco
Corresponding Certification: CCNP Voice
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Cisco 642-427 Exam Syllabus Topics:

SectionWeightObjectives
Advanced Features and Applications20%- Call Control Discovery (CCD) and SAF
- Media resources, voice quality, and QoS issues
- Extension Mobility Cross Cluster
CUCM and Registration Troubleshooting25%- CUCM service availability and database replication
- Endpoint and IP phone registration issues
- Gateway and trunk registration (SIP, H.323, MGCP)
Integrated Solution Troubleshooting10%- Unity Connection integration issues
- Presence and IM service troubleshooting
Call Setup and Dial Plan Troubleshooting25%- Globalized call routing and number transformation
- Dial plan, route patterns, partitions, CSS, translation rules
- Intrasite, intersite, and off-net call failures
Troubleshooting Methodology and Tools20%- Monitoring and diagnostic tools: RTMT, Serviceability, traces, logs, debugs
- Systematic troubleshooting approach

Cisco Troubleshooting Cisco Unified Communications v8.0 (TVOICE v8.0) Sample Questions:

1. A company has migrated to a Cisco Unified CallManager IP Telephony system and is now replacing the existing voice-mail system with a Cisco Unity voice-mail system. A small group of users has been established to test the new voice mail system. The users were able to initialize their mailboxes and record greetings; however, during testing the callers were sometimes unable to leave voice-mail messages for Cisco Unity users.
Which two issues could cause this problem? (Choose two.)

A) There is a mismatch in the MWI on/off numbers configured in Cisco Unified CallManager and Cisco Unity.
B) Cisco Unity is in a G.729 region and has not been configured to support G.729.
C) There is a mismatch in the number of ports configured in Cisco Unified CallManager and Cisco Unity.
D) The call transfer call handlers are not configured correctly.
E) The hunt group is hunting to Cisco Unity ports that have been dedicated for message notification.


2. Users are complaining that they hear echo when they place calls to the PSTN through the HQ_PSTN gateway. The voice port configuration for the gateway is as follows:
voice-port 1/0:23
input gain -1
output attenuation 1
echo-cancel erl worst-case 3
Why are users hearing echo?
Select the best response.

A) The echo canceller should be configured so that SIN and SOUT have a 2 dB difference in order to function properly.
B) InSignalLevel needs to be adjusted so that it is within the range from -20 dB to -25 dB
C) The actual ERL is 2dB; however, the echo canceller still does not acknowledge the input signal as echo because it is lower than the echo-cancel erl worst-case 3 command.
D) The echo canceller is functioning properly and some echo is to be expected on every call.


3. Please choose two methods of LRQ forwarding from the following items. (Select two.)

A) LRQ sequential
B) LRQ static
C) LRQ init
D) LRQ blast


4. As a network administrator, you should be familiar with the bandwidth command. Which parameter of the bandwidth command is used in gatekeeper configuration mode to specify the maximum amount of bandwidth that can be allocated in a zone?

A) total
B) remote
C) session
D) interzone


5. Refer to the exhibit.

When a Cisco IP Communicator phone roams from San Jose (SJ) to RTP, the Cisco IP Communicator physical location and the device mobility group change from SJ to RTP All route patterns are assigned a route list that points to the local route group All device pools are configured to use the local route group Which statement is true when the roaming phone places an AAR call?

A) The phone will use the AAR CSS that contains the SJ_PSTN partition. The call will egress at the SJ gateway
B) The phone will use the AAR CSS that contains the SJ_PSTN partition. The call will egress at the RTP gateway.
C) Since globalized call routing is not configured, then the SJ gateway will be used in this case
D) The phone will use the AAR CSS that contains the RTP_PSTN partition. The call will egress at the SJ gateway
E) The phone will use the AAR CSS that contains the RTP_PSTN partition The call will egress at the RTP gateway


Solutions:

Question # 1
Answer: A,E
Question # 2
Answer: C
Question # 3
Answer: A,D
Question # 4
Answer: A
Question # 5
Answer: B

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