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ONT – Optimizing Converged Cisco Networks : 642-845 Exam

Exam Number/Code: 642-845
Exam Name: ONT – Optimizing Converged Cisco Networks
VUE Code: 642-845
Questions Type: Multiple choice,
Exam Language(s): English

“ONT – Optimizing Converged Cisco Networks”, also known as 642-845 exam, is a Cisco certification.
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QUESTION 27:
Certkiller has several offices around the country. They have deployed a VOIP
network with Cisco Unified CallManager in a multisite centralized configuration
but there is a requirement to place a call agent at some of the remote sites. Which
IOS gateway feature should be deployed at the remote site to enable local call
control?
A. Survivable Remote Site Telephony (SRST)
B. Gatekeeper multizone
C. Cisco Unified CallManager Express (CME)
D. Call Admission Control (CAC)
E. Automatic Alternate Routing (AAR)
F. None of the above
Answer: C
Explanation:
Cisco Unified CallManager is the IP-based PBX in an IP telephony solution. It acts as a
call agent for IP phones and MGCP gateways and can also interact with H.323 or SIP
devices using their protocols. For redundancy and load sharing, multiple Cisco Unified
CallManager servers operate in a cluster. From an administration perspective, the whole
cluster is a single logical instance.
Cisco IOS routers can permanently act as a call agent for IP phones. The feature that
provides this functionality is Cisco Unified CallManager Express. With Cisco Unified
CallManager Express, Cisco Unified CallManager functionality is provided by the router.
If the router is also a voice gateway, it combines IP telephony and VoIP gateway
functionality in a single box.
Cisco IOS gateways also support other features, such as call preservation (Real-Time
Transport Protocol [RTP] stream) in case of a lost signaling channel, dual tone
multifrequency (DTMF) relay capabilities, supplementary services support (for user
functions, such as hold, transfer, and conferencing), and fax and modem support.
QUESTION 28:
Two Certkiller phones connect to Cisco Gateway routers as shown below:
Certkiller 1 configuration exhibit:
Certkiller 2 configuration exhibit:
Study the exhibits shown above. Both routers have been configured as VoIP
gateways. They must also support traditional telephony devices to connect to analog
telephones. Which two configuration changes would correctly support the voice
requirements? (Select two)
A. On each router, under the dial-peer voice 1 pots configuration, add the port 1/0/0
command.
B. Under the dial-peer voice 2 voip configuration, change the destination pattern of 1111
to 2222 on the Certkiller 1 router, and 2222 to 1111 on the Certkiller 2 router.
C. Under the dial-peer voice 2 voip configuration, change the destination target address
of 10.2.2.2 to 10.1.1.1 on the Certkiller 1 router, and the destination target address of
10.1.1.1 to 10.2.2.2 on the Certkiller 2 router.
D. On each router, configure dial-peer voice 1 as a voip connection and configure
dial-peer voice 2 as a pots connection.
E. On each router, under the dial-peer voice 1 pots configuration, add the port fa0/0
command.
F. Under the dial-peer voice 1 pots configuration, change the destination pattern of 1111
to 2222 on the Certkiller 1 router, and 2222 to 1111 on the Certkiller 2 router.
Answer: A, B
Explanation:
Command Description
dial-peer voicetag type Use the dial-peer voice command to
enter the dial peer subconfiguration
mode. The tag value is a number that
has to be unique for all dial peers
within the same gateway. The type
value indicates the type of dial peer
(for example, POTS or VoIP).
destination-pattern
telephone_number
The destination-pattern command,
entered in dial peer subconfiguration
mode, defines the telephone number
that applies to the dial peer. A call
placed to this number will be routed
according to the configuration type
and port (in the case of a POTS type
dial peer) or session target (in the
case of a VoIP type dial peer) of the
dial peer.
portport-number The port command, entered in POTS
dial peer subconfiguration mode,
defines the port number that applies
to the dial peer. Calls that are routed
using this dial peer are sent to the
specified port. The port command
can be configured only on a POTS
dial peer.
session target ipv4:ip-address The session target command, entered
in VoIP dial peer subconfiguration
mode, defines the IP address of the
target VoIP device that applies to the
dial peer. Calls that are routed using
this dial peer are sent to the specified
IP address. The session target
command can be configured only on
a VoIP dial peer.
QUESTION 29:
You need to determine the correct ports to use on a new Cisco router which will be
used as a VOIP gateway. Which two statements are true about voice ports on a
router? (Select two)
A. Calls made to the PSTN can be made via FXS or T1/E1 trunk ports.
B. Analog and IP phones can be connected to the VoIP network via FXO or T1/E1 trunk
ports.
C. Calls to the PSTN can be made via the use of FXO or T1/E1 trunk ports.
D. Calls between analog phones that are attached to the FXS ports in a VoIP network can
be completely processed by voice-enabled routers.
Answer: C, D

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