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Question 1.
What can you use to verify real-time call control processing in a VoIP network?
A. debug VoIP rtcp
B. debug call control
C. debug VoIP ccapi inout
D. debug voice call control
Answer: A
Explanation:
To enable debugging for Real-Time Transport Control Protocol (RTCP) packets, use the debug voip rtcp command in privileged EXEC mode.
The debug voip ccapi inout EXEC command traces the execution path through the call control API, which serves as the interface between the call session application and the underlying network-specific software. You can use the output from this command to understand how calls are being handled by the router.
This command shows how a call flows through the system. Using this debug level, you can see the call setup and teardown operations performed on both the telephony and network call legs.
Reference:
http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_command_reference_chapter09186a00801aa213.html
http://www.cisco.com/univercd/cc/td/doc/product/software/ios113ed/113t/113t_1/voip/debug.htm
Question 2.
A voice gateway is a box that ______.
A. Connects two dissimilar networks.
B. Transports voice and restricts data.
C. Can support only a distributed call processing model.
D. Cannot be connected to the traditional PSTN network.
Answer: B
Question 3.
You have a customer that operates a group of factories. Each factory has an analog phone at each location. These phones are connected to an FXS port on the on-site router.
The press operators are unable to make any phone calls from these analog phones.
Use the following output to resolve the problem:
2611#s voice port 1/0/0
Foreign Exchange Station 1/0/0 Slot is 1, Sub-unit is 0,
Port is 0
Type of Voice Port is FXS
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
Non Linear Mute is disabled
Non Linear Threshold is -21 dB
Music On Hold Threshold is Set to 38 dBm
In Gain is Set to 0 dB
Out Attention is Set to 3 dB
Echo Cancellation is enabled
Echo Cancellation NLP mute is disabled
Echo Cancellation NLP threshold is -21 dB
Echo Cancel Coverage is set to default
Play out-delay Mode is set to default
Play out-delay Nominal is set to 60 ms
Play out-delay Maximal is set to 200 ms
Play out-delay Minimum mode is set to default, value 40 ms
Play out-delay Fax is set to 300 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Call Disconnect Time Out is set to 60 s
Ringing Time Out is set to 180
Wait Release Time Out is set to 30 s
Companding Type is u-law
Region Tone is set for US
Analog Info Follows:
Currently processing none
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to 600r Ohm
Station name None, Station number None
Voice card specific Info Follows:
Signal Type is ground Start
Ring Frequency is 25 Hz
Hook Status is On Hook
Ring Active Status is inactive
Ring Ground Status is inactive
Tip Ground Status is inactive
Digit Duration Status is inactive
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
No disconnect acknowledge
Ring Cadence is defined by CPTone Selection
Ring Cadance are [20 40] * 100 msec
2611#
What is the cause of this problem?
A. Incorrect cptone
B. Incorrect dial-type
C. Incorrect signal type
D. Incorrect disconnect-ack
Answer: C
Question 4.
A company has been using the following dial peer codec command:
Codec g729r8
Over the weekend they reconfigured their dial peers with the following command:
Codec g729ar8 bytes 10
How does this affect their voice network bandwidth and delay characteristics?
A. There will be no change.
B. Delay will increase on a per call basis.
C. Delay will decrease on a per call basis.
D. Bandwidth consumption will decrease on a per call basis.
E. Bandwidth consumption will increase on a per call basis.
Answer: E
Question 5.
Which two features render VAD ineffective? (Choose two)
A. Fax
B. CNG
C. Call waiting
D. Music on hold
E. Call forwarding
Answer: A, D
Explanation:
Over time and as an average on a volume of more than 24 calls, VAD may provide up to a 35 percent bandwidth savings. The savings are not realized on every individual voice call, or on any specific point measurement. For the purposes of network design and bandwidth engineering, VAD should not be taken into account, especially on links that carry fewer than 24 voice calls simultaneously. Various features such as music on hold and fax render VAD ineffective. When the network is engineered for the full voice call bandwidth, all savings provided by VAD are available to data applications.
Reference:
http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml
Question 6.
A user is trying to call another user over a VoIP network and gets a busy tone instead of a dial tone.
What command should you use to troubleshoot the problem?
A. show voice dsp
B. show voice connection
C. show voice port summary
D. show dial-peer voice summary
Answer: A
Explanation:
To show the current status of all digital signal processor (DSP) voice channels, use the show voice dsp command in privileged EXEC mode.
To display configuration information about a specific voice port, use the show voice port command in EXEC mode.
order to check the validity of the dial peer configuration, use the Cisco IOS command show dial-peer voice summary.
Question 7.
What does compressed RTP significantly reduce?
A. Packet delay
B. Total bandwidth
C. Frame Relay overhead
D. Total number of packets
Answer: B
Explanation:
Compressed RTP (CRTP), specified in RFC 2509, was developed to decrease the size of the IP, UDP, and RTP headers.
Question 8.
While installing a voice gateway outside the United States, what two requirements need to be verified? (Choose two)
A. PSTN standards in that country.
B. Encryption capabilities legalities.
C. The service provider installing the gateway.
D. Supplementary service including fax and modem.
Answer: A, B
Question 9.
A customer in Great Britain needs to install a Cisco router to support IP Telephony services with direct-connected analog phones.
What FXS port parameter do you need to change to emulate the local PSTN provider?
A. Signal
B. Cptone
C. Busy out
D. Description
Answer: B
Question 10.
Your customer is a computer components warehouse. To keep costs low, all inside sales associates are located at corporate headquarters in another state.
Your customer is interested in providing a direct analog telephone connection to the inside sales teams from the pick-up counters at their warehouses. This connection will not require the customer to dial any digits.
One of the warehouses is having a problem with their sales phone.
Given the following output:
altwhse#show voice port 1/0:1
Foreign Exchange Office
Type of Voice Port is E&M
Operation State is DORMANT
Administrative State is UP
The Last Interface Down Failure Cause is Administrative Shutdown
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Connection Mode is plar
Connection Number is 2000
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Call-Disconnect Time Out is set to 60 s
Ringing Time Out is set to 180 s
Region Tone is set for US
What is causing the calls to fail?
A. Voice Port type is incorrect.
B. Echo cancellation is enabled.
C. Connection Number is not required.
D. Interdigit Time Out is set to 10 seconds.
Answer: A
Question 11.
Which three types of trunks does Cisco support with the connection trunk command?
(Choose three)
A. FXS to FXS trunks
B. FXS to FXO trunks
C. FXO to FXO trunks
D. E&M to FXS trunks
E. E&M to FXO trunks
F. E&M to E&M trunks
Answer: A, B, F
Question 12.
If no incoming dial peer matches a router or gateway, the incoming call leg _____.
A. Takes an alternate path.
B. Matches the default dial peer.
C. Sends a busy to the originator.
D. Is denied and the call does not complete.
Answer: B
Question 13.
The following configuration is used at Site A:
dial-peer voice 20 pots
destination-pattern 20
port 1.0:1
dial-peer voice 41 VoIP
destination-pattern 41
session target ipv4:10.2.0.20
The following configuration is used at site B:
dial-peer voice 40 pots
destination-pattern 41
port 1.0:1
dial-peer voice 20 VoIP
destination-pattern 20
session target ipv4:10.4.1.41
To configure a permanent connection between the PBXs, what must be added to the voice port configuration at site A?
A. connection trunk 20
B. connection trunk 41
C. connection tie-line 20
D. connection tie-line 41
Answer: B
Explanation:
You must specify the same number in the connection trunk voice port command as in the appropriate dial peer destination-pattern command in order to create a permanent trunk.
Question 14.
A telephony service provider sells managed IP Phone service to businesses in multi-tenant units.
The provider has POPs in many cities, so all of their dial peer patterns are based on 10 digit numbers. Users dial 9 for local calls, followed by the 7 digital local number.
In a Chicago POP, the following dial peer has been configured:
dial-peer voice 312 pots
destination-pattern 312
port 1/0:24
A user dials a local number, 9-555-0597.
What command must be configured in the gateway to allow the call to complete?
A. prefix 312
B. forward-digits 7
C. rule 1 9.......312.......
D. num-exp 9.......312.......
Answer: D
Question 15.
Which configuration defines a destination pattern for all of the 1000 and 2000 range of extensions starting with the numbers 555?
A. 5551...
B. 5552...
C. 555[1-2]...
D. 555[1000-2000]...
Answer: C
Question 16.
What does RTCP provide?
A. Independent services irrespective of RTP.
B. Compression techniques to save bandwidth.
C. In-band control information for an RTP flow.
D. Out-of-band control information for an RTP flow.
Answer: D
Explanation:
RTCP provides out-of-band control information for an RTP flow.
Question 17.
What is the disadvantage of using VoIP rather than VoFR or VoATM?
A. Data can arrive out of sequence.
B. Networks are complicated to design.
C. Data units can arrive out of sequence.
D. Network failures are not automatically found.
Answer: C
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Question 1.
LSC validation is failing on a new SCCP IP phone that you have just added to the Cisco Unified CallManager 5.0 cluster. No other IP phones are experiencing any problems with LSC validation.
What can you do to help pinpoint the problem?
A. Check for security alarms
B. Verify that the authentication string is correct in the Cisco Unified CallManager device
configuration screen
C. View the SDI trace output
D. Use the Security configuration menu on the IP phone to verify that an LSC has been
downloaded to the IP phone
Answer: D
Question 2.
Which three capabilities can't be configured if the default dial peer is matched? (Choose three.)
A. Set preference to 1
B. Invoke a Tcl application
C. Enable dtmf-relay
D. Set codec to G.711
E. Disable DID
F. Disable VAD
Answer: B, C, F
Question 3.
When using trace output to troubleshooting a Cisco Unified CallManager 5.0 problem, how can you collect and view the trace files?
A. Configure the proper trace settings on the Cisco Unified CallManager Serviceability page and
then use the embedded RTMT tool to view the trace files
B. Configure the proper trace settings on the Cisco Unified CallManager Serviceability page and
download the RTMT plug-in from the CallManager Administration page to view the trace output
C. Download the RTMT plug-in from the Cisco Unified CallManager Serviceability page to view
the preconfigured trace files
D. Configure the proper alarms and traces on the Cisco Unified CallManager Administration page
and view the output with the RTMT plug-in
Answer: B
Question 4.
You are troubleshooting why a user can't make calls to the PSTN. You are reviewing trace files and you found where the user's IP phone initiates the call but you never see the call go out the gateway.
What is the next step in troubleshooting this issue?
A. Look in the SDL trace file to see if there is a signal to another Cisco Unified CallManager node
with the same time-stamp
B. Look in the MGCP trace file to determine which MGCP gateway the call was sent to
C. Look in the IP Voice Media Streaming APP trace file to see if an MTP was invoked
D. Look in the SDL trace file to see if there is a signal to anther Cisco Unified CallManager node
with the same TCP handle
Answer: A
Question 5.
Exhibit:
Your work as a network engineer at ITCertKeys.com. Please study the exhibit carefully.
Voice bearer traffic is mapped to which queue in FastEthernet0/2?
A. Queue 2
B. Queue 3
C. Queue 1
D. Queue 4
Answer: A
Question 6.
Exhibit:
Your work as a network engineer at ITCertKeys.com. Please study the exhibit carefully. You have received a trouble ticket stating that calls to international numbers are failing. To place an international call, users dial the access code, "9," followed by "011", the country code and the destination number. After entering the debug isdn 1931 command on the MGCP gateway, you have the user attempt the call again.
Base on the debug output, what is the most likely cause of this problem?
A. The TON in incorrect
B. The circuit is not configured correctly or has a physical layer issue
C. Cisco Unified CallManaager is not stripping the access code before sending the call to the
gateway
D. The gateway dial peer needs to prefix "011" to the called number so the PSTN knows this is
an internation call
E. The user's CSS does not permit international calls
Answer: A
Question 7.
You have developed a dial plan for Cisco Unified CallManager 5.0 solution. All the route patterns, partitions, calling search spaces and translation rules have been configured. Before starting up the system you wish to test the dial plan for errors.
Which Cisco Unified CallManager tool will simplify this testing?
A. Route Plan Report
B. Dial Plan Installer
C. Dialed Number Analyzer
D. RTMT Traces and Alarms
Answer: C
Question 8.
You have just obtained a list of the following options:
all patterns
unassigned DN
Call Park
Conference
Directory Number
Translation Pattern
Call pickup group
Route pattern
Message waiting
Voice mail
Attendant console
What have you selected in order to produce this list?
A. Route Plan > External Route Plan Wizard
B. Control Center > Feature Services
C. Route Plan > Route Plan Report
D. Dialed Number Analyzer
Answer: C
Question 9.
You have configured the Enable Keep Alive check under Trace Filter settings.
How does this change the trace output?
A. It adds the IP address of the endpoint in hex
B. It maps the unique TCP handle for the endpoint to the MAC address of the endpoint in the
trace output
C. It adds the SCCP messages and all fields sent as part of that message
D. It adds TCP socket numbers between the endpoint and Cisco Unified CallManager for the
session
Answer: B
Question 10.
When using trace output to troubleshooting a Cisco Unified CallManager 5.0 problem, how can you collect and view the trace files?
A. Configure the proper trace settings on the Cisco Unified CallManager Serviceability page and
download the RTMT plug-in from the CallManager Administration page to view the trace output
B. Configure the proper alarms and traces on the Cisco Unified CallManager Administration page
and view the output with the RTMT plug-in
C. Download the RTMT plug-in from the Cisco Unified CallManager Serviceability page to view
the preconfigured trace files
D. Configure the proper trace settings on the Cisco Unified CallManager Serviceability page and
then use the embedded RTMT tool to view the trace files
Answer: A
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