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Latest CCNP Voice 642-437 Dumps (11-15)

Exhibit See Full Version.
The one thing that distinguishes a PRI from other interfaces is the fact that the data that is received from the PSTN on the D-channel and needs to be carried in its raw form back to the Cisco CallManager to be processed. The gateway does not process or change this signalling data, it simply passes it onto the Cisco CallManager through TCP port 2428. The gateway is still responsible for the termination of the Layer 2 data. That means that all the Q.921 data-link layer connection protocols are terminated on the gateway, but everything above that (Q.931 network layer data and beyond) is passed onto the Cisco CallManager. This also means that the gateway does not bring up the D-channel unless it can communicate with Cisco CallManager to backhaul the Q.931 messages contained in the D-channel. http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a00801da84e.shtml

A new Cisco 7965 IP phone is installed on a Cisco Unified Communications Manager Express
system. When the phone requests the .loads file from the TFTP server, it sees that the versions are different. What does the IP phone do to resolve this issue?
A. The IP phone requests the SEP<mac>.cfg file and reboots.
B. The IP phone attempts to obtain the new firmware file image from the TFTP server.
C. The IP phone boot requests the XMLDefault.cnf.xml file and boots up.
D. The IP phone does not boot up and will require manual intervention to factory reset the phone before a new firmware image can be downloaded.
Answer: B
Explanation: Cisco IP Phone Initialization Process:
At initialization, the Cisco IP phone sends a request to the DHCP server to get an IP address, DNS server address, and TFTP server name or address, if appropriate. Options are set in DHCP server (Option 066, Option 150, and so on). It also gets a default gateway address if set in DHCP server (Option 003).
If a DNS name of the TFTP sever is sent by DHCP, then a DNS sever IP address is required to map the name to an IP address. This step is bypassed if the DHCP server sends the IP address of the TFTP server. In this case study, the DHCP server sent the IP address of TFTP because DNS was not configured.
If a TFTP server name is not included in the DHCP reply, then the Cisco IP phone uses the default server name.
The configuration file (.cnf) file is retrieved from the TFTP server. All .cnf files have the name SEP<mac_address>.cnf, where "SEP" is an acronym for Selsius Ethernet Phone. If this is the first time the phone is registering with the Cisco CallManager, then a default file, SEPdefault.cnf, is downloaded to the Cisco IP phone.
All .cnf files include the IP address(es) of the primary and secondary Cisco CallManager(s). The Cisco IP phone uses the IP address to contact the primary Cisco CallManager and register. 6.Once the Cisco IP phone has connected and registered with Cisco CallManager, the Cisco CallManager tells the Cisco IP phone which executable version (called a load ID) to run. If the specified version does not match the executing version on the Cisco IP phone, the Cisco IP phone will request the new executable from the TFTP server and reset automatically. http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080129d92. shtml

You are configuring a network to support voice to the PSTN. One important aspect to the configuration is to be able to determine the individual slot, subunit, and port number from the gateway endpoint identifier. Which signaling protocol is appropriate for this situation?
A. H.323
Answer: D
Explanation: Endpoints are any of the voice ports on the designated gateway. These voice ports provide connectivity to both analog ports and digital trunks to the PSTN. Ports on gateways are identified by endpoints in very specific ways. It is important to note that gateways can have multiple endpoints dependent on the number of ports it contains, and that the endpoints are case insensitive.

Exhibit See Full Version.
Inbound Dial Peers Matching Process
When the Cisco IOS router or gateway receives a call setup request, a dial peer match is made for the incoming call in order to facilitate routing the call to different session applications. This is not a digit-by-digit match; rather the full digit string received in the setup request is used to match against configured dial peers. The router or gateway matches the information elements in the setup message with the dial peer attributes to select an inbound dial peer. The router or gateway matches these items in this order:
Called number (DNIS) with the incoming called-number command:
First, the router or gateway attempts to match the called number of the call setup request with the configured incoming called-number of each dial peer. Because call setups always include DNIS information, it is recommended to use the incoming called-number command for inbound dial peer matching. This attribute has matching priority over the answer-address and destination-pattern commands.
Calling Number (ANI) with the answer-address command:
If no match is found in step 1, the router or gateway attempts to match the calling number of the call setup request with the answer-address of each dial peer. This attribute can be useful in situations where you want to match calls based on the calling number (originating).
Calling Number (ANI) with the destination-pattern command:
If no match is found in step 2, the router or gateway attempts to match the calling number of the call setup request to the destination-pattern of each dial peer. For more information about this, see the first bullet in the Dial Peer Additional Information section of this document.
Voice-port (associated with the incoming call setup request) with configured dial peer port (applicable for inbound POTS call legs):
If no match is found in the step 3, the router or gateway attempts to match the configured dial peer port to the voice-port associated with the incoming call. If multiple dial peers have the same port configured, the dial peer first added in the configuration is matched.
If no match is found in the first four steps, then the default dial peer 0 command is used.
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml#co nv

How many IP phone calls can be sent across a 64-kb/s Frame Relay link that uses the G.729 codec? The sampling rate is 50 times a second, with 20 bytes per sample. There are 8 bytes of Frame Relay header overhead with no checksum, and header compression is used.
A. 3
B. 4
C. 5
D. 7
Answer: C
Explanation: Bandwidth Calculation FormulasThese calculations are used:http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.sh tml