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Cisco 350-030

CCIE Voice Exam Version: 4.29

Topic 1, Volume A QUESTION NO: 1 On which gateway or gatekeeper is the IOS command call-rsvp-sync resvtimer 10 used to set the timer? A. originating VoIP gateway for completing RSVP reservation setups in 10 seconds B. originating and terminating VoIP gateway for completing RSVP reservation setups in 10 seconds C. terminating VoIP gateway for completing RSVP reservation setups in 10 seconds D. VoIP gatekeeper for completing RSVP reservation setups in 10 seconds Answer: C QUESTION NO: 2 Calls to an ICD queue should reserve an available agent and connect the call after a database lookup is performed. How should the script be configured to accomplish this? A. Set the Resource Step Connect option to No and perform a Connect after the database lookup is completed. B. Issue a Call Hold after the Resource Step selects an agent and release the hold after the database lookup is completed. C. Issue a Queue Step followed by the database lookup and a Resource Step. D. Issue a Queue Step followed by the database lookup and a Dequeue Step. Answer: A QUESTION NO: 3 What occurs if the system clocks are not synchronized between the sender and receiver of an RTP stream? A. Packets can be placed in sequence but jitter cannot be compensated for. B. Packets cannot be reordered, because sequence and jitter cannot be compensated for. C. Jitter can be compensated for, but packets cannot be reordered if

they arrive out of sequence. D. Packets may be reordered and jitter may be compensated for, because the timestamp is not related to the system time. E. When the RTP stream is opened, the sender and receiver synchronize their clocks before the stream commences so that packet sequencing and dejitter will function correctly. Answer: D

it will be redirected to the PSTN due to a lack of MTP resources C. one for the branch office. A total of two regions are required for all sites with G. The Cisco IP Voice Media Streaming App needs to be configured for G. E.711 codec.711 codec. Which of the following configurations would meet this requirement? A.729 codec between the two regions. and an n + 1 connection is attempted.711 because the phones are negotiating G. G. The MoH server should belong to the headquarters region.729 codec during call setup. C.729 codec between the two regions. The MoH server cannot be configured to transmit G. G. B.QUESTION NO: 4 If all n MTP transcoding sessions are utilized. Calls between the headquarters and the branch office utilize the G. Multicast MoH must always be transmitted using the G. and one for the MoH server.729 codec specified between the two regions. it will use the alternate codec type and attempt to complete the call D. Calls within both the headquarters and the branch office utilize the G.711 codec is used within regions.711 codec is used within regions.729 codec.711 codec utilized between itself and each office. G. Three separate regions are required: one for the headquarters. A total of two regions are required for all sites with G. D.711 codec is used within regions. A total of two regions are required for all sites with G. Codecs between and within each region are . it will complete the call without using the MTP transcoding resource Answer: D QUESTION NO: 5 A centralized call processing topology comprises a headquarters and a branch office. it will not use an MTP and will use the transcoding resources associated with the codec to complete the ca B.711 codec only. The MoH server is placed in a separate location with the G. how will the next call be treated? A.

6 kb/s. cRTP on: 24 kb/s QUESTION NO: 7 Acme Widgets Inc. cRTP D.0 kb/s B. 20 percent Answer: E QUESTION NO: 9 .specified accordingly. It is currently using G. Answer: E QUESTION NO: 6 On a WAN PPP link.2 kb/s. 9.711 VoIP call packetized at 50 p/s? A.729. cRTP on: 33. cRTP on: 11. cRTP B. cRTP C. implements cRTP using the ip rtp header-compression command. cRTP Answer: off: off: off: off: off: D 72. what will be the Layer 3 bandwidth consumption per call on the WAN link? A.729 VoIP calls when cRTP is turned off. wants to compress the voice data traveling over its WAN connection to its parent company.2 kb/s 48 kb/s.0 kb/s E. 50 percent C. what is the required bandwidth for three G. 16.6 kb/s 90 kb/s. 8. loading two voice frames per packet.8 kb/s C. cRTP on: 36 kb/s 79. 30 percent E. 60 percent B. 12.6 kb/s 26.) A. 8.6 kb/s D. cRTP E. When Acme Widgets Inc.4 kb/s. cRTP on: 24. and what is it when cRTP is turned on? (Note The payload size is 20 bytes. 40 percent D.0 kb/s Answer: A QUESTION NO: 8 Approximately what percentage of overall bandwidth is saved (at Layer 3) by cRTP for a G.

The 4-byte header has 2 bytes of padding in order to align the packet on a word boundary. and 20-byte IP header) can be reduced to either 2 or 4 bytes. 72000*1000* C. C. The 4-byte cRTP packet contains an additional field containing the full destination IP address. The 4-byte cRTP packet contains a UDP Checksum.) A. The 4-byte cRTP packet contains an additional field containing the full source IP address. 62000*1000* Answer: A QUESTION NO: 10 When compressing RTP traffic across low-speed links using the cRTP feature. and therefore the IP phones are in SRST mode and have registered to the gateway with the configuration in the exhibit. 21000* E.Refer to the exhibit. Based on the configuration shown. what digit pattern will the voice-mail server see if there is no answer when IP phone B is called from phone A? (Note: Assume that the Cisco Unified Communications Manager servers have become unreachable. What is the primary difference between the 2-byte cRTP packet and the 4-byte cRTP packet? A. 52000*1000* B. the 40 bytes of header information (12-byte RTP header. 42000*1000* D. The 4-byte header is used for larger payload RTP packets (such as 30 ms samples. D. E. as opposed to 10 ms samples) Answer: C QUESTION NO: 11 When implementing PRI backhaul for an MGCP gateway and Cisco Unified Communications . 8-byte UDP header. B.

Manager. Cisco CTI Manager closes the provider while calls at JTAPI ports and route points that have not yet been terminated get redirected to the Call Forward On Failure number that has been configured for them. Audit Connection (AUCX) F. Cisco CTI Manager and Cisco TSP provide both TAPI and JTAPI applications with access to specific Cisco Unified Communications Manager servers in a cluster. Make Connection (MKCX) C. Modify Connection (MDCX) D. such as an IP phone Answer: B QUESTION NO: 12 Which two of these are invalid MGCP commands? A. Cisco CTI Manager and Cisco TSP provide TAPI applications with access to Cisco Unified Communications Manager resources and functionality without being aware of specific Cisco Unified Communications Manager systems. signaling link terminal D. Delete Connection (DLCX) E.921 data-link protocol is terminated on which device? A. B.F QUESTION NO: 13 Which function does Cisco CTI Manager provide for Cisco Unified Communications Manager clusters? A. D. C. the IP end device. Answer: A QUESTION NO: 14 . Reset In Progress (REIP) Answer: B. Cisco Unified Communications Manager B. When a Cisco TAPI application fails. When a Cisco Unified Communications Manager node fails. Notification Request (RQNT) B. Cisco CTI Manager recovers the affected JTAPI ports and route points. MGCP gateway C. the Q.

proxy B. are altered with additional information. SIP Phone B requests that Server B place its calls for it. After the source host. 802. 802. Following the VLAN header is the actual ether type for the frame and length information. and 12 bits for the VLAN identifier. with 16 bits for the ether type (0x8100). What kind of device is Server B? A. 802. Following the VLAN header is the actual ether type for the frame and length information. and 12 bits for the VLAN identifier. which normally contain the destination host.10 frame headers. a 48bit 802. source host. user agent client E.10 frame headers. After the source host. and protocol. with 32 bits for the ether type (0x8000).1Q header is included.1Q header is included.1Q header is included.10 frame headers.1Q header is included. registrar D. and 12 bits for the VLAN identifier. source host.1Q frame header? A. and protocol. 1 bit for the canonical field (always 0). 3 bits for the priority field (not used in this implementation). are altered with additional information. redirect C. After the source host. which normally contain the destination host. and protocol. 1 bit for the canonical field (always 0). a 16bit 802. with 16 bits for the ether type (0x8100). 3 bits for the priority field (not used in this implementation). which normally contain the destination host. a 24bit 802. 3 bits for the priority field (not used in this implementation).10 frame headers. source host. which normally contain the destination host.To hide its identity when initiating calls. 802. Following the VLAN header is the actual ether type for the frame and length information. . C. After the source host. with 16 bits for the ether type (0x8000). 3 bits for the priority field (not used in this implementation). source host. user agent server Answer: A QUESTION NO: 15 Which of these is true about the 802. 1 bit for the canonical field (always 0). B. and protocol. are altered with additional information. a 32bit 802. D. are altered with additional information.

1 bit for the canonical field (always 0). Following the VLAN . and 12 bits for the VLAN identifier.

cnf file from TFTP server -send StationRegister Skinny Protocol message to Cisco Unified Communications Manager -send StationAlarm Skinny Protocol message to Cisco Unified Communications Manager -download phone button template and softkeys from Cisco Unified Communications Manager B.) A.cnf file from TFTP server -send StationAlarm Skinny Protocol message to Cisco Unified Communications Manager -send StationRegister Skinny Protocol message to Cisco Unified Communications Manager -download phone button template and softkeys from Cisco Unified Communications Manager D. .download SEP(MAC). Guaranteed Quality of Service C. Answer: B QUESTION NO: 16 Which of these features are supported in RSVP Support for LLQ? (Choose three. Controlled-Load Network Element Service Answer: B.cnf file from TFTP server -download local files from TFTP server -download phone button template and softkeys from Cisco Unified Communications Manager -send StationAlarm Skinny Protocol message to Cisco Unified Communications Manager -send StationRegister Skinny Protocol message to Cisco Unified Communications Manager C.download SEP(MAC). . LLO Support on Tunnels B.cnf file from TFTP server -download local files from TFTP server . Reserve resources for Low Latency and bandwidth guarantees D. as seen in the sniffer traces. that takes place when an IP phone is initially registering with Cisco Unified Communications Manager? A.C. LLQ on Frame Relay and ATM PVCs E.download local files from TFTP server -download SEP(MAC).download local files from TFTP server -download SEP(MAC).E QUESTION NO: 17 Which of these is the correct sequence of events.header is the actual ether type for the frame and length information. . .

.cnf file from TFTP server -send StationAlarm Skinny Protocol message to Cisco Unified Communications Manager -send StationRegister Skinny Protocol message to Cisco Unified Communications Manager -download phone button template and softkeys from Cisco Unified Communications Manager .-send StationAlarm Skinny Protocol message to Cisco Unified Communications Manager -send StationRegister Skinny Protocol message to Cisco Unified Communications Manager -download phone button template and softkeys from Cisco Unified Communications Manager E.download SEP(MAC).

D. Connect B. QueueDelay D. Select Resource Answer: D QUESTION NO: 19 Which three attributes correctly describe aspects of SIP? (Choose three. centralized dial plan management F. Which step in the ICD palette will offer a call to an Event Service Desk? A. peer-to-peer B. intelligent endpoints Answer: A.-download local files from TFTP server Answer: D QUESTION NO: 18 Refer to the exhibit. Master/Slave C. communication with Cisco Unified Communications Manager handled via a proxy server E.F .) A. Dequeue C. call preservation on gateway failover from one Cisco Unified Communications Manager server to another D.

Call Transmit . policy-map BULK-2RATE-3COLOR-MARKER class BULK police cir 5000000 pir 10000000 conform-action transmit exceed-action set-dscp-transmit 12 violate-action set-dscp-transmit 14 C. policy-map BULK-2RATE-3COLOR-MARKER class BULK police conform-rate 5000000 violate-rate 10000000 conform-action transmit exceed-action set-dscp-transmit afl2 violate-action set-dscp-transmit afl3 E. Ring Off B. On Hook C. policy-map BULK-2RATE-3COLOR-MARKER class BULK police car 5000000 be 5000 conform-action set-dscp-transmit afll exceed-action set-dscp-transmit afl2 violate-action set-dscp-transmit afl3 B. threecolor marker that is compliant with RFC 2597. policy-map BULK-2RATE-3COLOR-MARKER class BULK police cir 5000000 burst 5000000 conform-action transmit exceed-action set-dscp-transmit afl2 violate-action drop D. policy-map BULK-2RATE-3COLOR-MARKER class BULK police cir 5000000 pir 10000000 conform-action set-dscp-transmit 11 exceed-action set-dscp-transmit 12 violate-action set-dscp-transmit 13 Answer: B QUESTION NO: 21 Which three options are valid SCCP call states sent to an IP phone? A. and the OoS Baseline to police bulk traffic to a conforming rate of 5 Mb/s and a violating rate of 10 Mb/s? A. RFC 2698.QUESTION NO: 20 Which configuration produces the best design for a two-rate.

323 C. What information will the server return to the caller? A.D. which Protocol will satisfy the following requirements? Requirement 1: the protocol has a mechanism for a centralized dialplan Requirement 2: the endpoints are considered to be unintelligent Requirement 3: the protocol is text-based A.323. Connected E. You are debugging a problem on a SIP network and have run the debug ccsip messages command. H. SIP B. In Use Remotely Answer: B. the acceptable media type B. MGCP and SCCP in a VOIP deployment. H. a list of acceptable media types C. One of the messages returned is shown in the exhibit. Disconnected F.D.F QUESTION NO: 22 When comparing SIP. SCCP Answer: C QUESTION NO: 23 Refer to the exhibit. a list of acceptable formats . MGCP D.

Drag the existing Menu step and drop it on Output 3 of the new Menu. or transfer to the operator. You have been asked to edit the sample auto attendant script so that callers are prompted to press 1 for sales.D. dial by name. 2 for service. B. Drag a new Menu step from the palette and drop it on the existing Menu step. . they should hear the existing menu choices to dial by extension. or 3 for the directory. This will make the existing Menu subordinate to the new Menu. an acceptable language code Answer: C QUESTION NO: 24 Refer to the exhibit. What steps can you take to create this nested menu? A. If callers select 3. a correct directory number E. Drag a new Menu step from the palette and drop it on the Start step.

E. Drag the existing Menu step and drop it on Output 3 of the new D. step from the palette and directory menu as the Which three of these are mandatory sub-commands of the call-managerfallback command and will help an IP phone register to an IOS router in SRST mode? (Choose three. Delete the existing Menu. . ip source-address D. access-code B. dialplan-pattern C. The gatekeeper configurations are shown as entered by the administrator. max-ephones Answer: C.C. Drag a new Menu drop it on the Set prefixPrompt=P[] step. Recreate the existing third option of the new Menu step. Drag a new Menu step from the palette and Menu step. max-dn F. keepalive E.F QUESTION NO: 26 Refer to the exhibit.) A. Answer: C QUESTION NO: 25 drop it on the existing Menu.

From the Call Admission Control perspective of Cisco Unified Communications Manager. B.) A. This design requires that Call Admission Control be performed on the central site link independently of the branch links.The gatekeepers are not functioning correctly. a service-provider IP WAN service that is based on MPLS is. D. Which three of these are possible problems? (Choose three. The Cisco Unified Communications Manager servers need to be aware of the underlying MPLS network by setting the appropriate enterprise parameters.E QUESTION NO: 28 Cadorna Inc. in reality.D QUESTION NO: 27 Which two of the following are functions of DHCP snooping? (Choose two. relies on already discovered trusted and untmsted ports B. each site is two hops away from all other sites.B. E. From an IP routing perspective on the enterprise side of the network.) A. They are interconnected with an MPLS network that provides full-mesh connectivity between all sites. The zone prefix commands are incorrect. equivalent to a hub-andspoke topology without a hub site. The gatekeeper functions have not been activated. automatically builds ACLs Answer: C. is planning a deployment of Cisco IP telephony using the centralized call processing model. The bandwidth commands are incorrect. D. E. dynamic ARP inspection C. Which two statements are true? (Choose two. builds a binding table F. uses existing binding tables E. The company has a headquarters location with 25 branches. C. The gw-type-prefix commands are incorrect. B. defines trusted and untrusted ports D.) A. This design using MPLS will not allow Call Admission Control to be . C. The zone local and zone remote commands are incorrect Answer: A.

Answer: B.C .performed.

It uses a multicast packet with a destination MAC address of 01-00CC-CC-CC. Answer: A. B. voice flows generated from third-party applications. voice flows generated from Cisco IOS applications C.E QUESTION NO: 32 Which three statements apply to the route filter function in Cisco Unified Communications . 66 kb/s C. 55. A. C. 60. all traffic marked DSCP EF E.D.8 kb/s D. The default is 60 seconds. The platform TLV (TLV type 0x0006) contains an ASCII character string that describes the hardware platform of the device E. F.) A. D. all RSVP bandwidth requests B.9 kb/s Answer: D QUESTION NO: 31 If enabled. It is an excellent tool for displaying the interface status on switches. 72 kb/s B.QUESTION NO: 29 Which three statements are true about Cisco Discovery Protocol? (Choose three.E QUESTION NO: 30 What is the required bandwidth for three G. It works on top of the network layer and data link level. the RSVP for LLQ feature will assign which two types of flows to the priority queue? (Choose two.) A. all traffic marked CoS 5 Answer: C.729 VoIP calls on a WAN Frame Relay link with cRTP turned off? (Note: The payload size is 30 bytes. such as Microsoft NetMeeting D. It uses a broadcast packet with a destination MAC address of 01-00CC-CC-CC. You can use the CDP timer feature to change update times.

Manager? (Choose three.) .

managers and assistants have separate directory numbers and calls are usually diverted to assistants' lines. switchport priority extend cos 0 C. You would like the IP phone connected to the switch port in voice VLAN to assign a Layer 2 priority to all packets coming from the PC to default 0.E QUESTION NO: 33 An IP phone is connected to a Cisco Inline Power switch port. The switch port is acting as a trunk and is running both voice and data VLAN configuration on it. B. managers and assistants have separate directory numbers or lines on their phones. but calls to managers are usually diverted to assistants' lines. A route filter can be used to restrict dialing certain numbers. B. E. A route filter can be used to identify a subset of a wildcard pattern.A.B. switchport trunk priority cos 0 D. C. A route filter cannot effectively block the dialing of 900 area codes. The switch is running Native IOS image. A route filter cannot be used in conjunction with partitions and Calling Search Spaces to set up complex dialing rules Answer: A." wildcard denotes a portion of a route pattern that is not stripped when the pattern matches. Which IOS CLI in the interface port configuration for the inline power switch will help you achieve this objective? A. switchport access priority extend cos 0 B. In shared-line mode. switchport mode access priority extend cos 0 E. The ". switchport access extend cos 0 Answer: B QUESTION NO: 34 How does the function of IPMA in proxy-line mode differ from IPMA in shared-line mode? A. The wildcard "@" denotes any number that you can dial in the North American numbering system. In proxy-line mode. F. D. mis qos priority extend cos 0 F. .

managers and assistants have separate directory numbers. but calls to managers are usually diverted to assistants' lines.C. In proxy-line mode. In shared-line mode. Answer: A . but assistants can handle calls without disturbing managers D. managers and assistants have separate directory numbers or lines on their phones.

C. It defines which numbers are available for a device to call. B. It defines the search for directory numbers in assigned partitions according to dial patterns. C. A referenced CTI Route Point is not associated with the JTAPI user. B. D.QUESTION NO: 35 Which two descriptions apply to the Calling Search Space function in Cisco Unified Communications Manager? (Choose two. each CSS has a directory number. It defines route patterns and directory numbers from which calls can be received.D.) A. the new music files are in the correct format to be used with Cisco Unified Communications Manager E. Within a partition. From the trace. There is an error in one of the scripts being loaded. Which three of these do you need to verify? (Choose three.E QUESTION NO: 36 Users are complaining that the music on hold marketing files for this month are not being played when users are placed on hold. the IP voice media streaming application has been stopped and restarted B. E.E QUESTION NO: 37 Refer to the exhibit. users have selected the correct MoH files for customer calls D. The JTAPI user password is not correct. D.) A. Answer: A. It provides a group of dial patterns to look through when making a call. a new directory has been created for the new media files C. E. the location of the new music files is what the MoH server expects Answer: A. . The JTAPI Subsystem is showing OUT_OF_SERVICE. what is the cause of this issue? A. The CTI Manager service is not running on Cisco Unified Communications Manager. Cisco Unified Contact Center is not able to resolve the host name of Cisco Unified Communications Manager.

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729 (G. 6xx Answer: D QUESTION NO: 40 Which two of the following functions best match and describe a Cisco IOS Software MTP? (Choose two. 5xx E. but only .) A. It transcodes between G.729ab) codecs. 64 bytes B. as seen by the sniffer? A.711 codecs. It transcodes a-law to mu-law and vice versa for G. 204 bytes F. 4xx D.729 and G. 160 bytes D. 128btyes C. 3xx C. 1 xx B.711 (a-law.711 codecs.711 with a 20 ms sample.729a. mu-law) and G.729b. C. B.Answer: A QUESTION NO: 38 An IP phone configured in voice VLAN 128 is registered to a Cisco Unified Communications Manager cluster. What will be the data frame size coming out of the phone. It supports the G. 200 bytes E. G. The codec configured on both of the phones is G. G. 218 byte Answer: F QUESTION NO: 39 Which type of SIP responses would indicate that a server encountered an error in attempting to complete a SIP request? A. A PC is connected to the back of an IP phone which is running the sniffer program and collecting packets coming out of the IP phone. It has a speech connection established with another IP phone that is connected to another Cisco Unified Communications Manager cluster.

one codec is supported at any one time .

Answer: A.) A. communication with Cisco Unified Communications Manager handled via a proxy server E. 240 kb/s C. Master/Slave C. 192 kb/s B.C QUESTION NO: 41 What is the required bandwidth for three G.5 kb/s D. Cisco IOS Software MTPs are identical to the transcoder-type Cisco IOS Media Termination Point. 60 50 40 30 20 percent percent percent percent percent . C.8 kb/s Answer: C QUESTION NO: 42 Which three attributes correctly describe aspects of MGCP? (Choose three.E QUESTION NO: 43 Approximately what percentage of overall bandwidth is saved (at Layer 2) by cRTP for a G. B. peer-to-peer B. intelligent endpoints Answer: B.D. 223.711 VoIP calls on an Ethernet link? (Note that the packets-per-second count should be set to 33 p/s. centralized dial plan management F. A.C. MTPs are used to extend supplementary services to H323 devices that support Empty Capability Set. D.729 VoIP call packetized at 50 p/s and running over a MLP link? A. E. E. call preservation on gateway failover from one Cisco Unified Communications Manager server to another D. 238.

A site CTI Route Point has been configured such that calls to extension 5900 are forwarded to Cisco Unity for call handler greeting administration. the administrator hears the opening greeting instead of Cisco Unity Greeting Administrator What is the probable cause? A. A greeting has not been recorded for mailbx 5900 Answer: B QUESTION NO: 45 Which of these is a pair of RSVP reservation types? A. The call routing rule is not configured correctly. Extension 5900 has not been defined in Cisco Unity. When an administrator calls extension 5900. The voice-mail box in Cisco Unified Communications Manager is set to 1020 instead of XXXX. C.Answer: B QUESTION NO: 44 Refer to the exhibit. Reservation and Path Answer: A QUESTION NO: 46 . Distinct and Shared B. B. Same and Distinct C. Shared and non-shared D. D.

) a third-party fax gateway. C.) A. a software conference bridge that is configured in Cisco Unified Communications Manager and a HW transcoder C. a hardware conference bridge D. The fax originated from protocol-based Cisco fax relay.729 codec.C QUESTION NO: 48 Which two statements apply to the partitions function in Cisco Unified Communications Manager? (Choose two. No extra configurations required-phones automatically negotiate using the lowest common denominator codec (G. calls within each site are using the G. The fax originated from protocol-based fax passthrough. The fax originated from B.711 codec. The fax originated from NTE-based fax passthrough.Refer to the exhibit.) A. D. Answer: C. a software conference bridge that is configured in Cisco Unified Communications Manager B. Which two of these could be the failure reasons? (Choose two. The debug outputs that are shown in the exhibit were collected at the terminating Cisco IOS gateway for a fax call that failed. Calls between the two sites are using the G. which two of the following are possible solutions? (Choose two. a Cisco gateway that is configured with a a Cisco gateway that is configured with a a Cisco gateway that is configured with an .729) Answer: B. To conference an existing call between two phones at the central site with a phone at the remote office. a hardware transcoder and a hardware conference bridge E.D QUESTION NO: 47 Company Alpha has a central office and a branch office that utilize a central call processing topology.

A. A partition is a logical grouping of directory numbers and route patterns that have similar reachability characteristics. each CSS has a directory number. C. D. When a directory number or route pattern is placed into a certain partition. this creates a rule for who can call that device or route list B. E. A directory number may appear in only one partition. Calling Search Spaces are assigned to partitions. Within the partition. .

Answer: A. Assume that you need to limit the bandwidth (to one G.711 call) between IP phones and the voice gateway (psy-gw-01). SIP Notify C. SIP INFO D. gatekeeper . SIP Subscribe/Notify Answer: B QUESTION NO: 50 Refer to the exhibit. RTP-NTE B.323v3 standard. This gatekeeper (psy-gk-01) conforms to the H. in-band audio E.B QUESTION NO: 49 What is the default DTMF relay for Cisco Unity Express when integrated via SIP? A. What will be the correct gatekeeper configuration (psygk-01)? A.

89. when a user listens to only part of a message and hangs up .89.228 zone prefix horse 2* gw-priority 10 10.228 zone prefix horse 2* gw-priority 10 10.89.129.com 10.226 gw-type-prefix 1#* default-technology bandwidth total zone horse 64 no shutdown B.89.89. when a user deletes a new message without listening to it C.211 zone prefix horse 2* gw-priority 0 10.com 10.129.129. gatekeeper zone local horse maui-onions.129.226 gw-type-prefix 1#* default-technology bandwidth total zone horse 128 no shutdown Answer: E Explanation: Topic 2.226 gw-type-prefix 1#* default-technology bandwidth total zone horse 160 no shutdown E.89.228 zone prefix horse 2* gw-priority 10 10.129.89.com 10.89.129. gatekeeper zone local horse maui-onions.com 10.89.89.211 zone prefix horse 2* gw-priority 0 10.129. gatekeeper zone local horse maui-onions.211 zone prefix horse 2* gw-priority 0 10. gatekeeper zone local horse maui-onions.228 zone prefix horse 2* gw-priority 10 10.228 zone prefix horse 2* gw-priority 10 10.com 10.129.89.226 gw-type-prefix 1#* default-technology bandwidth total zone horse 84 no shutdown D. when all new messages are listened to B.89.226 gw-type-prefix 1#* default-technology bandwidth total zone horse 80 no shutdown C.211 zone prefix horse 2* gw-priority 0 10.129.zone local horse maui-onions.89.129.89.211 zone prefix horse 2* gw-priority 0 10.129.89.129. Volume B QUESTION NO: 51 Which event does not trigger Cisco Unity Connection to turn off MWI for a user IP phone? A.129.129.129.

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5 C. 12 Answer: E QUESTION NO: 53 . How many simultaneous G.729 calls can be established between sites SJ and RTP? A. 4 B. 6 D. 8 E.D. none of the above Answer: E QUESTION NO: 52 Refer to the exhibit. when a user deletes a message after listening to only part of it E.

What will the status of the call be if a resource is available when the Resource Step is executed? A. Selected D. Queued E. Queued for Connection Answer: C QUESTION NO: 54 Which digital telephony signaling protocol does not support ANI information? A.Refer to the exhibit. Connected B. T1 PRI . Queued for Selection C.

To forward a call to a mailbox. and the redirecting number. vm-integration pattern direct#cgn* pattern ext-to-ext no-answer # cgn * cdn pattern ext-to-ext busy # cgn * cdn . vm-integration pattern direct 7500 # cgn * pattern ext-to-ext no-answer 7500 # cgn * fdn pattern ext-to-ext busy 7500 # cgn * fdn pattern trunk-to-ext no-answer 7500 # cgn * fdn pattern trunk-to-ext busy 7500 # cgn * fdn B. T1 CAS E&M Feature Group D Answer: B QUESTION NO: 55 Refer to the exhibit.B. To prompt a caller to enter a password to access his or her mailbox. vm-integration pattern direct 7500 # cgn * pattern ext-to-ext no-answer 7500 # cgn * cdn pattern ext-to-ext busy 7500 # cgn * cdn pattern trunk-to-ext no-answer 7500 # cgn * cdn pattern trunk-to-ext busy 7500 # cgn * cdn C. an "*". T1 BRI E. Which vm-integration configuration should be added to the branch gateway to support voicemail while in SRST mode? A. the legacy voicemail system at headquarters expects to receive a "#" followed by the calling number. the voicemail system expects to receive a "*" followed by the calling number and a "#". T1 CAS E&M Feature Group B C. E1 R2 D.

B QUESTION NO: 57 . vm-integration pattern direct#cgn* pattern ext-to-ext no-answer # cgn * fdn pattern ext-to-ext busy # cgn * fdn pattern trunk-to-ext no-answer # cgn * fdn pattern trunk-to-ext busy # cgn * fdn Answer: D QUESTION NO: 56 Refer to the exhibit. The Standard CTI Enabled group is not added to the Cisco Unified Presence user in Cisco Unified Communications Manager. The log was captured for a Cisco Unified Presence client that is not able to perform desk phone control to a Cisco IP phone.pattern trunk-to-ext no-answer # cgn * cdn pattern trunk-to-ext busy # cgn * cdn D. The directory number of the IP phone is not configured with "Allow Control of Device from CTI." D. Answer: A. The IP phone is not configured with "Allow Control of Device from CTI.) A. The IP phone is not registered." C. Which two of these could be the potential causes that are revealed by the log? (Choose two. B.

Selected D. Queue Immediate B. Connected E.Refer to the exhibit. Queued for Selection F. Queued for Connection Answer: D QUESTION NO: 58 A customer needs to create a plan for deploying QoS across the company network in anticipation of adding a VoIP solution and to clean up the existing network. Queued C. What will the status of the call be if a resource is available when the Resource Step is executed? A. Employees currently use a .

D.D. Which four CoS-DSCP values should be applied to the customer network so that voice. E. video. Once a media resource is associated with a Media Resource Group. E&M signaling semicompelled signaling loop-start signaling line signaling Group 1 signaling . The Media Resource Group configuration page allows administrators to choose whether to use multicast for MOH audio. voice signaling CS3 Answer: A. A Media Resource Group contains a prioritized list of media resources. which has been only marginally successful because of poor quality. The default Media Resource Group is defined in the service parameters of Cisco Unified Communications Manager. ERP and manufacturing protocols AF31 B. Different types of media resources cannot be grouped into the same Media Resource Group. video signaling CS3 C. B. video AF41. Internet BE E. C. the Intemet for customer support and customer orders. and data all function properly? (Choose four. E. Answer: E QUESTION NO: 60 Which three are valid T1 CAS types? (Choose three. Voice bearer AF43. and a recently installed IP-based videoconferencing system. B. D. voice signaling AF26 D.corporate intranet to report expenses and access ERP and manufacturing applications. ERP and manufacturing protocols AF43 F. it is no longer eligible to be associated with another Media Resource Group.) A.) A. C.B. voice bearer EF.F QUESTION NO: 59 Which statement about the Media Resource Group on Cisco Unified Communications Manager is correct? A.

ground-start signaling Answer: A.F.C.F .

A referenced CTI Route Point is not associated with the JTAPI user. TFTP file manipulation E.) A. B. Answer: B. phone and server identity theft D.C QUESTION NO: 62 Integrity. E. man-in-the-middle attacks C. B. There is an error in one of the scripts being loaded. modification of call-processing signals between Cisco Unified Communications Manager and IP phones B. E. unauthorized capturing and decoding of voice signaling packets Answer: D QUESTION NO: 63 Which of the following three messages could be sent by the UAC in response to the 180 Ringing? (Choose three. authentication. Which of these describes an integrity threat? A. The CTI Manager service is not running on Cisco Unified Communications Manager.) A. The JTAPI user password is not correct. and encryption are some of the security features that are supported on Cisco Unified Communications Manager and IP phones. C.QUESTION NO: 61 Which two of these are possible reasons why a JTAPI subsystem might have the status PARTIAL_SERVICE? (Choose two. C. D. D. Cisco Unified Contact Center is not able to resolve the host name of Cisco Unified Communications Manager. PR ACK ACK BYE CANCEL INVITE .

ground-start signaling E.Answer: A.38 fax call? A. would be negotiated for a T. tone-based supervisory disconnect C. 24000 E. 19200 Answer: C QUESTION NO: 66 .D QUESTION NO: 64 Which signaling method cannot solve the FXO disconnect problem? A. in bits per second. Which data rate. 28800 C. battery reversal Answer: C QUESTION NO: 65 Refer to the exhibit.C. 14400 D. 33600 B. pulse dial D. power denial B.

analog FXS E.D QUESTION NO: 67 Which two telephony interfaces on a Cisco IOS gateway can be controlled by Cisco Unified Communications Manager using SCCP? (Choose two.) A. analog FXO Answer: B.D QUESTION NO: 68 Refer to the exhibit.Refer to the exhibit. SW1 egress port E. If the trust boundary has been extended to the IP phone on SW1. SW1 ingress port C. IP phone attached to SW1 B. analog E&M B. ISDN BRI C. in what two places will traffic be marked and classified so that the proper QoS settings may be carried through the network? (Choose two. Traffic flows from the IP phone that is connected to SW1 to the IP phone on SW2. ISDN PRI D. Which gatekeeper mechanism prevents the gatekeeper from using all the resources on either gateway 1 or gateway 2 when sending calls to zones SE and NW? . R1 egress port Answer: B. R1 ingress port D.) A.

2728 E. classification B. You are working with your security department. ras timeout brq Answer: C QUESTION NO: 69 LLQ belongs to which Cisco quality of service feature? A. 2427 B. bandwidth total D. congestion management C. After the firewall is turned up. 9900 Answer: B QUESTION NO: 71 Which statement about the H. You suspect that the backhaul port has been blocked. 2428 C. bandwidth remote B. bandwidth zone E. your PRI is no longer working. resource availability indicator C. which has a firewall in place between Cisco Unified Communications Manager and the gateway. congestion avoidance D. Irq immediate advance F. What port does Cisco Unified Communications Manager use between itself and an MGCP gateway for PRI backhaul traffic? A.323 Fast Connect (also known as fast start) procedures is correct? . 2727 D. shaping and policing E. link efficiency mechanisms Answer: B QUESTION NO: 70 You are implementing a Cisco Unified Communications Manager solution utilizing MGCP gateways with PRI interface cards.A.

C. B. Fast Connect shortens the call setup time by allowing H. setup time by allowing H. IGMP snooping does not examine or snoop Layer 3 information in packets that are sent between the hosts and the router. An IP multicast stream to the IP host can be stopped only by an IGMP leave message. D. exchanges.A. never C. D. Answer: A QUESTION NO: 72 Which of these is not a valid switchback method for SCCP hardware conference bridges? A.323 devices to propose and confirm master-slave determination in H.323 devices to use UDP as a transport protocol for H.225 setup to H. Fast Connect shortens the call reclaiming time that is otherwise used for TCP handshakes.225 exchanges. E. guard E. IGMP control messages are transmitted as IGMP multicast packets so that they can be . When a host in a multicast group sends an IGMP leave message. C. B.225 message connect. only that port is deleted from the multicast group. E.245 negotiation messages.225 steps and move directly from H. uptime Answer: B QUESTION NO: 73 Which three statements are true about multicast IGMP snooping? (Choose three.) A.323 devices to propose and confirm Open Logical Channel in H. immediate B. Fast Connect shortens the call setup time by permitting calls to bypass certain H.245 negotiations to take place earlier during the H. Fast Connect shortens the call setup time by allowing H.225 message exchanges. the switch adds the host's port number to the associated multicast table entry. graceful D. When the switch hears the IGMP host report from a host for a particular multicast group. Fast Connect shortens the call setup time by forcing all H.225 message exchanges.

distinguished from normal multicast data at Layer 2.C. F. Answer: A. A switch that is running IGMP snooping examines every multicast data packet to verify whether it contains any pertinent IGMP "must control" information.D .

QUESTION NO: 74 What are two advantages of multicast technologies? (Choose two. CTI ports B. Answer: D. B. which reduces bandwidth D. allows instant messages D. C. They control network traffic and reduce server and CPU load. Assume UDP checksum is enabled.729 codec bit rate is 8 kb/s. AXL service C.729 call crossing a Frame Relay link when ip rtp header-compression is enabled. . filters incoming presence status requests Answer: B QUESTION NO: 77 Calculate the percentage of bandwidth that is saved for one G. The Frame Relay overhead is 4 bytes. dialog groups F.) A.E QUESTION NO: 76 Which of these best describes the Incoming ACL" configuration on Cisco Unified Presence? A. E. They eliminate multipoint applications. The G. bypasses digest authentication C. Denial of service attacks in the network are prevented. allows incoming certificates to Cisco Unified Presence E.E QUESTION NO: 75 Which two Cisco Unified Contact Center Express system components do not support integration redundancy? (Choose two. CSQ E. They eliminate traffic redundancy. They reduce traffic by delivering a separate stream of information to each corporate recipient or home environment. Cisco Unified CM Telephony trigger D. HTTP trigger Answer: D.) A. permits incoming packets to Cisco Unified Presence B.

bits E. 3.2 2. approximately D. bits D. 7 E 31 15. 6. approximately B. In this IPv4 packet. approximately C. approximately E. 7 6. Which of the policies will effectively provision for IP telephony traffic being carried over a 768 kb/s Frame Relay PVC using Cisco recommended best-practice parameters? . which bits in the ToS byte are used for ECN? A. time slots 1 E.4 5. Bits Answer: 0. and 17 to 31 to 32 to 15. and 17 to 31 15. timeslots to C. timeslots to Answer: D QUESTION NO: 79 Refer to the exhibit. time slots 1 D. bits B. approximately Answer: D QUESTION NO: 78 55 85 20 65 60 percent percent percent percent percent Which E1 time slots are used to carry encoded voice only? A. timeslots to B.1.1 0. and 17 to 32 QUESTION NO: 80 These three-class policies use MQC tools whenever possible. bits C.A.

policy-map WAN-EDGE class VOICE priority percent 33 compress header ip rtp class CALL-SIGNALING bandwidth percent 5 ! policy-map HQC-FRTS-768 class class-default shape average 729600 7296 0 fragment frfl2 960 service-policy WAN-EDGE ! . policy-map WAN-EDGE class VOICE priority percent 33 class CALL-SIGNALING bandwidth percent 5 ! policy-map HQC-FRTS-768 class class-default shape average 768000 7680 0 service-policy WAN-EDGE ! interface Serial2/0.12 point-to-point frame-relay interface-dlci 102 class FR-HAP-CLASS-768 ! map-class frame-relay FR-HAP-CLASS-768 service-policy output HQC-FRTS-768 C.A. policy-map WAN-EDGE class VOICE priority percent 33 compress header ip rtp class CALL-SIGNALING bandwidth percent 5 ! policy-map HQC-FRTS-768 class class-default shape average 729600 7296 0 service-policy WAN-EDGE ! interface Serial2/0.12 point-to-point frame-relay interface-dlci 102 class FR-HAP-CLASS-768 map-class frame-relay FR-HAP-CLASS-768 service-policy output HQC-FRTS-768 frame-relay fragment 960 B.

policy-map WAN-EDGE class VOICE priority percent 70 class CALL-SIGNALING bandwidth percent 5 ! policy-map HQC-FRTS-768 class class-default shape average 729600 7296 0 service-policy WAN-EDGE ! interface Serial2/0. .interface Serial2/0. policy-map WAN-EDGE class VOICE priority percent 33 class CALL-SIGNALING bandwidth percent 5 ! policy-map HQC-FRTS-768 class class-default shape average 729600 7296 0 compress header ip rtp service-policy WAN-EDGE ! interface Serial2/0.12 point-to-point frame-relay interface-dlci 102 ip rtp header-compression class FR-HAP-CLASS-768 ! map-class frame-relay FR-HAP-CLASS-768 service-policy output HQC-FRTS-768 frame-relay fragment 960 E.12 point-to-point frame-relay interface-dlci 102 class FR-HAP-CLASS-768 D.12 point-to-point frame-relay interface-dlci 102 class FR-HAP-CLASS-768 ! map-class frame-relay FR-HAP-CLASS-768 service-policy output HQC-FRTS-768 frame-relay fragment 640 Answer: E Explanation: The correct answer must be E as there is map-class configured which inherits the policy-map and also fragmentation must be defined inside Frame Relay map-class.

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711 codec with VAD disabled. instructing it to switch to the G. lower than B. the fax tones sent by the contacting fax machine cause the OGW to send an NSF message to the TGW. lower than C. + C. lower than E. ? Answer: E QUESTION NO: 82 How is fax pass-through traffic treated over IP WAN connections that use the G. D.QUESTION NO: 81 Which Cisco Unified Communications Manager route pattern character represents zero or more occurrences of the previous digit or wildcard? A. the TGW changes to the G. The fax traffic is demodulated and sent with VAD and echo chancellor disabled.711 codec with echo chancellor and VAD disabled. * D. lower than Answer: D or or or or or equal equal equal equal equal to to to to to 10 Mb/s 384 kb/s 1.544 Mb/s 768 kb/s 512 kb/s . . C. cRTP is recommended on which link speed? A. When the TGW detects the CED tone from the fax machine that has been contacted. ! B. B. When the CFR message is received. the OGW is informed by the contacted device of the Cisco NSF features and switches to the G.711 codec with echo chancellor and VAD disabled. lower than D.729 codec? A. Answer: D QUESTION NO: 83 According to the Cisco QoS SRND guide. The contacting fax machine sends a TCF message to the contacted fax machine and waits for a CFR message. When the OGW detects the CED tone from the fax machine that is making the call. E.

QUESTION NO: 84 .

Answer: B. Assuming that the IP phone uses SCCP. VAD is enabled.An IP phone user just answered an incoming call by lifting the handset. Station C. Station E.711 mu-law codec is used to transport modem tones. C. which SCCP message will Cisco Unified Communications Manager transmit to this called IP phone immediately after receiving notification about the off-hook event? A. Clear-channel codec is used to transport modem tones. D. E.C. B. Station D. G. Station B. F NLP is enabled.) A. VAD is disabled.E QUESTION NO: 86 . NLP is disabled. Station Answer: C Media Port List message Set Ringer message Stop Tone message Start Media Transmission message Open Receive Channel message QUESTION NO: 85 Which three statements about a modem pass-through call are correct? (Choose three.

Refer to the exhibit. What is the maximum number of inbound calls to 2001 before a Cisco Unified Communications Manager Express system returns a user busy tone to any additional calls? A. 3 B. 4 C. 5 D. 6 E. 7 Answer: E QUESTION NO: 87 You are deploying a new Cisco Unified Communications solution that utilizes a centralized call processing model. Which method of CAC is recommended for this type of solution? A. RSVP-based B. gatekeeper-based C. CUBE-based D. locations-based Answer: D QUESTION NO: 88 Which Cisco IOS CLI command can be used to identify the high jitter level of an RTP stream on a Cisco IOS voice gateway? A. show B. show C. show D. show E. show Answer: call active voice brief voip rtp connections voice dsp detailed voice call summary policy-map interface A

QUESTION NO: 89

Refer to the exhibit. This Cisco Unified Communications Manager trace shows a SIP message that is sent by a SIP Cisco Unified IP Phone 7965 to Cisco Unified Communications Manager. Which of these regarding the content of this SIP message is correct? A. phone registration message to the primary Cisco Unified Communications Manager B. keepalive message to the primary Cisco Unified Communications Manager C. phone registration message to the secondary Cisco Unified Communications Manager during a server failover D. keepalive message to the secondary Cisco Unified Communications Manager E. phone registration message to the primary Cisco Unified Communications Manager during fallback Answer: B QUESTION NO: 90 Which two analog voice interfaces support ground-start? (Choose two.) A. FXS B. E&MTypel C. E&MTypell D. E&MTypelV E. FXO Answer: A,E QUESTION NO: 91 When using DNS, what does the TC bit indicate? A. The total response size was greater than 512 bytes, and the data that does not fit was required B. The DNS response was less than 512 bytes, and the data required padding. C. The packet was greater than 1024 bytes, and the data was possibly damaged.

D. The pointer to the RRset that will provide the correct data due to possible damage Answer: A QUESTION NO: 92 Which Cisco Unified Presence service parameter must be modified from the default value in order for presence and instant messaging to be functional? A. server name B. server IP address C. DNS domain D. SIP proxy domain E. enable presence Answer: D QUESTION NO: 93 On a Cisco IOS MGCP PRI gateway, what is the maximum configurable length of time for a scheduled switchback to a higher-priority Cisco Unified Communications Manager? A. 6 hours B. 12 hours C. 18 hours D. 24 hours E. 48 hours Answer: D QUESTION NO: 94 What will occur if a video bridge is not available during a video call setup on Cisco Unified Communications Manager using the Intelligent Bridge Selection feature? A. The video B. The video Manager will video image. C. The video D. The video E. The video Answer: D call will fail, and the caller will receive a busy tone. call will succeed, and Cisco Unified Communications use the default call will fail and will be forwarded to voice mail. call will fall back to an audio-only call. call will fail, and the caller will hear a message.

H. H.245 signal C. The audio stream and audio server used will be selected according to the configuration of the phone being placed on hold. The audio stream will be selected according to the configuration of the phone being placed on hold and the audio server used will be selected according to the . H. The audio stream will be selected according to the configuration of the phone which is being used to place a caller on hold.) A.QUESTION NO: 95 Your client has a business requirement that mandates exact DTMF durations being passed endtoend across an H. in-band voice Answer: C. C. 14 MB E. and the audio server used will be selected according to the configuration of the phone being placed on hold. RTP-NTE E. 2 MB B. The audio stream and audio server used will be selected according to the configuration of the phone which is being used to place a caller on hold.323 VoIP infrastructure. Cisco RTP B. D. 10 MB D.245 alphanumeric D. 4MB C.D QUESTION NO: 96 What is the default mailbox size that triggers disablement of sending and receiving voice messages for a Cisco Unity Connection user? A. B.225 Notify F. 20 MB Answer: D QUESTION NO: 97 Which of these statements correctly describes the logic for selecting MoH servers and MoH audio streams? A. Which two DTMF relay methods meet the client requirement? (Choose two.

Answer: C .configuration of the phone which is being used to place a caller on hold.

III.I . CRCX D. IV B.III. RTPX C. from highest to lowest priority? A.I C. II.IV. When an inbound call with a calling number of 1001 and a called number of 2112 arrives at a Cisco IOS router with these dial peers. which MGCP message is used by Cisco Unified Communications Manager to inform the gateway where to send RTP traffic? A. what is the correct order of dial-peer matching. MDCX B.IV. RQNT Answer: A QUESTION NO: 99 Refer to the exhibit.II. NTFY E. II.QUESTION NO: 98 When a call is being established on a Cisco IOS MGCP gateway that is registered to Cisco Unified Communications Manager. III.

II. III.D. III. IV. I Answer: D QUESTION NO: 100 . IV. II.I E.

PPS. PPS. EOP. PPS. PPR.) A. DCN DCN DCN DCN Topic 3. NSF.000 phone license units for a Cisco Unified Communications Manager cluster. which two PSTN signals are processed locally by the gateway? (Choose two. EOP. EOP. Phone license unit overdrafts are never permitted Answer: B QUESTION NO: 103 Which statement about MRGL on Cisco Unified Communications Manager is incorrect? (Choose .Refer to the exhibit. PPS. EOP. EOP. Volume C QUESTION NO: 101 On a Cisco IOS MGCP gateway with a voice ISDN PRI connection to the PSTN. PPS. ISDNQ. T1 framing and line code C. PPR. NSF. PPS.30 message sequence will result in a successful fax transmission? A. Answer: A Explanation: PPR. MGCP endpoint configuration B. EOP. How many phone license unit overdrafts are permitted in this Cisco Unified Communications Manager cluster? A. MPS. 500 C. E.931 E. 700 D.30 message exchanges that resulted in a single page fax call failure. ISDN call setup Answer: B. DCN EOP. C. ISDNQ. 1000 E. D. Which T. The exhibit shows the T. NSF. PPS. EOP. B.921 D. RTP. MCF. PPS. MCF.C QUESTION NO: 102 A customer purchased 10. 200 B. EOP.

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Coder delay is the time that is taken to fill a packet payload with encoded/compressed speech B. Media resources that are not contained in any Media Resource Groups are not used by MRGL.E QUESTION NO: 104 Which three services must be activated on Cisco Unified Presence in order for presence and instant messaging to be functional? (Choose three.D. device pool level. D. When a call is placed on hold.B.three) A. E. Cisco Unified Presence Sync Agent F. E. Cisco Bulk Provisioning Service D. D. the MRGL of the device that put the call on hold determines which MOH server is used to play music to the held device.D QUESTION NO: 105 Which statement about coder delay in a VoIP network is correct? A. MRGL can contain a single Media Resource Group. Answer: D QUESTION NO: 106 When implementing a Cisco Unified Communications Manager solution over an MPLS WAN. C. or both. B. RSVP will transparently pass application IDs from the customer network across the MPLS . MRGL contains a prioritized list of Media Resource Groups. Cisco AXL Web Service C. which two rules must be observed to prevent overrunning the priority queue? (Choose two. Coder delay compensates for network switching delay. Coder delay varies with the voice coder that is used and the processor speed. MRGL can be assigned to devices at the device level. Answer: C. Coder delay transforms a variable delay into a fixed delay. Cisco Serviceability Reporter Answer: A. C. Coder delay is also known as algorithmic delay. Cisco Unified Presence Engine E.) A.) A. Cisco Unified Presence SIP Proxy B.

.WAN.

The media has to be symmetrically routed.) . C. it may support either topology-aware or measurementbased CAC. Subscriber Absent D. If call waiting is disabled for this directory number and none of the Call Forward settings are defined. which H. B. Network Busy Answer: D Explanation: QUESTION NO: 108 Which three statements are correct with regard to the Certificate Trust List file? (Choose three. C. User Busy E. No Route To Destination B. It contains identity. Only the connection to the MPLS WAN where the Cisco Unified Communications Manager resides must be enabled as a CE device. The media streams must be the same size in both directions. Answer: A.225 disconnect reason code will be sent to the originating H.B. If the CE is under corporate control. public key.C. It is a list of devices and credentials that a phone should trust on the network.D QUESTION NO: 107 An H. It is signed using the administrator workstation password. Normal Call Clearing C. D. Answer: B.225 call setup arrives at Cisco Unified Communications Manager for a directory number on an IP phone that is engaged in an active conversation.D Explanation: QUESTION NO: 109 What should you do to make good requirements for hardening an MCS operating system and its services? (Choose three.) A. D. E. It is created by the CTL client on the administrator workstation.323 gateway? A. and role information.

com. Existing web server and application logic can be used for VoiceXML voice applications. B. IMTs with bearer channels D. it indicates what? (Choose two. Keep the operating system up to date and install Cisco operating system upgrades that can be downloaded from Cisco. Answer: A. fast forward. B.D Explanation: QUESTION NO: 111 In Cisco SS7 Interconnect-based networks. D. and append are supported. which type of link from the PSTN can be determined by Signaling Link Terminal? A.B. Harden the IP stack by mitigating redirection attacks and enabling SYN flood protection.C.D Explanation: QUESTION NO: 110 Which three benefits are of using Cisco IOS VoiceXML? (Choose three. A-Link only Answer: B Explanation: QUESTION NO: 112 If the response to a DNS query comes back with a TC bit set. T1 with IMTs B. rewind.A.) . C. Use an intrusion protection system like Cisco Security Agent. HTML web development skills can be transferred to rose developing voice applications. Answer: A.) A. D. A-Link or F-Link C. Full editing features such as pause. C. Enable all services except file sharing. Hosting of VoiceXML voice applications can be added to the services that are offered to customers.

3 C. 4 D.711 conferences and 32 transcoding resources with an NM-HDV2? A. 615551234 C. how many DSPs will be required to support 8 G. Answer: A. 2 B. C. 61234 Answer: D Explanation: . B. 5 Answer: D Explanation: QUESTION NO: 114 When dialing 4085551234.*\( A. The size of the response exceeded 512 bytes. 611234 D. D. 614085551234 B. so only the first 1024 bytes were returned by the server. The size of the response exceeded 1024 bytes. so only the first 512 bytes were returned by the server.A. The size of the response exceeded 1500 bytes. The message was longer than the bandwidth would allow. what will be the output according to the following voice translation rule? Voice translation-rule 7 Rule 1/^.D Explanation: QUESTION NO: 113 If a DSP farm is configured in a centralized call-processing environment. so only the first 1500 bytes were returned by the server.

B. Psec tunneling in Windows 2000 of the Cisco Unified CallManager server Answer: A. B. IOS router configured in MGCP C. E.1 with a CTL installed B. copy the template and rename it. modify the template and update the changes. Cisco Unified CallManager 4. TCP TCP UDP UDP UDP 2000 2427 2427 2427 2427 and TCP 2002 and UDP 2428 and UDP 2428 and TCP 2428 .D Explanation: QUESTION NO: 116 Which of the following statements outline the correct way to implement a non-standard softkey template? A. C. Answer: A Explanation: QUESTION NO: 117 Which port(s) must be opened on an IOS firewall to allow successful MGCP (Media Gateway Control Protocol) message exchanges between a CallManager and an IOS MGCP PRI gateway? A. update it. modify it and update the changes. name it. B. modify the template and update the changes. Select add softkey template. D. modify it and update the changes. rename it. as well as SRTP between the gateway and IP phones? (Choose three.) A.QUESTION NO: 115 Which three requirements must be satisfied in order to enable signaling encryption between Cisco Unified CallManager and a gateway. Select the default softkey template. Select a softkey template. C. insert it. rename it. insert it. D. router with NM-HDV module D. Select a softkey template.

Cisco Unity Express and the CME gateway at each site may NOT be collocated in the same router chassis providing a voicemail access to local IP phones registered to local CME. D. C. Which way can CUE be deployed? A. F. One Cisco Unity Express can be used at the main site with CME router providing voicemail access to all the 3 sites. Phones need to trust all entries in the CTL file which could be CCM. B. Each site has a CME router with many IP phones in an IPT deployment. Answer: B. B. Another Cisco Unity Express with CME can be used at one of the remote sites to provide voicemail access for all of the IP phones at the two remote sites. The Network Administrator wants to provide all of the phones voicemail access using CUE. One Cisco Unity Express with CME can be used at the main site to provide voicemail access to the IP phones at the main site. Answer: A Explanation: QUESTION NO: 119 Which of the following are NOT true statements about Certificate Trust List (CTL) File? (Choose 2) A. It is a list of devices and credentials that a phone should trust on the network. CAPF. public key and role information. The CTL file is loaded to the phone each time when authentication is required. C. etc.Answer: E Explanation: QUESTION NO: 118 There are 2 remote sites and one main site. It contains identity. E. Cisco Unity Express and the CME gateway at each site must be collocated in the same router chassis providing voicemail access to local IP phones registered to local CME. TFTP. The CTL is created by CTL Client on administrator workstation.E Explanation: QUESTION NO: 120 . D. The CTL file is signed by administrator workstation password.

A CallManager Group can provide which two features to your call processing system? (Choose 2) .

MGCP D. Enables you to distribute the control of devices across multiple Cisco CallManagers C. SCCP Answer: C Explanation: QUESTION NO: 122 Which 2 are NOT functions performed by Cisco Media Streaming App Service? A.323.323 C. D. which Protocol will satisfy the following requirements: Requirement 1: It has a mechanism for a centralized dial-plan Requirement 2: The endpoints are considered to be unintelligent Requirement 3: The protocol must be text-based A. Provides SCCP stack for 4 software devices: ANN. H. B. Support for SRST in remote offices B. and SCCP in a VoIP deployment. CFB. and MGCP. MOH. Support for control of IPMA across primary and backup Cisco CallManagers for each group Answer: B. event logs. SIP B. and MTP Supports DB change notification processing Converts new MOH source files to separate WAV files for MOH codecs Provides SDI trace. H. and Perfmon counters Adjusts volume levels of MOH source files . Enables you to distribute voice mail support across multiple Unity servers D.D Explanation: QUESTION NO: 121 When comparing SIP.A. Support for redundancy by enabling you to designate a primary and backup Cisco CallManagers for each group E. C. E.

The route pattern strips the access code and site code and routes the call to the remote office's gateway. Regions D. Which of the following describes how these inter-office access codes should be configured? A. The translation pattern strips the access code and site code and is assigned a Calling Search Space that includes only the phones located in the office. A translation pattern is created for each office and is placed in a partition available to the phones at that office. A translation pattern is created for each office and placed in a partition available to all phones. C. All the remote offices have extensions in the range of 1000-1150. B. Locations C. To allow inter-office calls each office has been assigned a 3 digit site code. A route pattern is created for each office and placed in a partition available to all phones.F. D. The translation pattern strips the access code and site code and is assigned to Calling Search Space that includes all local phones. users will dial an access code followed by the 3 digit site code and the extension. A route pattern is created for each office and placed in a partition available to phones at that office. Answer: B Explanation: QUESTION NO: 124 Which type of media resources would be required for a single site call processing model? A. MTP B. MOH Answer: C. Transcoders Answer: A . To call between sites. The route pattern strips the access code and the site code and routes the call to the remote office's gateway. Provides audio data from WAV files: ANN.E Explanation: QUESTION NO: 123 A company has a headquarters with a centralized CallManager and 5 remote offices.

Explanation: .

Review all CCM User logs C. Which of the following . Calls from Division A to Division B are made using a site code of "919" followed by the recipient's 4 digit extension. All delivery locations get the same dialing domain ID as the primary location of the box they are created on. B. like ring settings reverting to default values D. Multiple primary location objects with the same value for this ID make up a dialing domain. Answer: C Explanation: QUESTION NO: 127 Two divisions in your company need to exchange Unity voice messages using VPIM. Dialing domains are also necessary if the Unity servers involved don't have overlapping dial plans. Look at the physical memory available of the server.QUESTION NO: 125 Which method could be used to determine if there is a JTAPI memory leak in a CallManager server? A. Determine if dialing the voice mail pilot number fails to connect to voice mail E. Check for changes to IP phone settings. Dialing domain IDs are stored on the primary location object. The primary extension in Unity is the user's four digit extension. All users in the dialing domain should be able to pick up their phones and dial each other directly without having to dial trunk access codes or use outside lines. C. Check for an increasing number of fast busys when dialing to the PSTN Answer: A Explanation: QUESTION NO: 126 Which statement does NOT describe dialing domain functionality in Cisco Unity? A. E. D. Dialing domains are multiple Unity servers that are handling subscribers that are on a single switch or networked switch B.

configurations on Division A's Unity server will allow messages to be forwarded between Unity systems using the same seven digit dialing that is used to place direct calls? (Choose 3) .

Performs digit manipulation D.F Explanation: both share the same directory number have separate directory numbers (DN). but each have separate directory numbers share a directory number (DN) and an IP . Matches dialed number for external calls C. The manager and assistant (DN). Points to the actual IP phone B. D. The manager and assistant (DN). C. Configure the recipient's four extension C.E Explanation: QUESTION NO: 128 digit number as an alternate extension to each VPIM Extension on the VPIM subscriber to "919" plus the digit Remote Mailbox Number on the VPIM subscriber to the digit Remote Phone Prefix to "919" Dial ID to "919" When configuring IP Manager Assistant (IPMA) in a shared line mode. The manager and assistant share an IPMA directory number. how are the manager and assistant Directory Numbers (DN) configured? A. Points to prioritized route groups Answer: A. Configure the Answer: B. The manager and assistant Manager Assistant (IPMA) directory number. Points to a route list for routing E. Chooses path for call routing F. B.C. Configure the E. Add the seven subscriber B.E. Answer: A Explanation: QUESTION NO: 129 Which 3 functions are NOT performed by a Route Pattern? (Choose 3) A.A. Configure the recipient's four extension D.

Enhanced Efficiency: controls network traffic and reduces server and CPU loads B. C. B. Answer: B. E. Terminating VoIP gateway for completing RSVP reservation setups .C. The TCP congestion control mechanisms decreases the congestion window when packet losses are detected ("slow start"). Distributed Applications: makes multipoint applications possible D. TCP does not contain the necessary timestamp and encoding information needed by the receiving application. Originating VoIP gateway for completing RSVP reservation setups within 10 seconds B. Bandwidth-conserving technology that reduces traffic by simultaneously delivering a single stream of information to thousands of corporate recipients and homes E. Prevent Denial of service (DoS) attacks in the networks Answer: E Explanation: QUESTION NO: 131 Why can't TCP be used for transferring audio and video over UDP? (Choose 5) A. TCP does not have a mechanism for sufficiently long buffering and adequate average throughput.D. Originating and terminating VoIP gateway for completing RSVP reservation setups within 10 seconds C. F.F Explanation: QUESTION NO: 132 The IOS command "call rsvp-sync resv-timer 10" is used to set the timer on the: A. TCP cannot support multicast. TCP headers are larger than a UDP header. D.QUESTION NO: 130 Which one of the following does NOT state Multicast Technologies Advantages? A. Optimized Performance: eliminates traffic redundancy C.E. Reliable transmission is inappropriate for delay-sensitive data such as real-time audio and video.

within 10 seconds .

Fake Answer A GetDigit Timeout value A call park watchdog timer Configure the interface for wink-start signaling 60% 50% 40% 30% 20% .D. VoIP gatekeeper for completing RSVP reservation setups within 10 seconds Answer: C Explanation: QUESTION NO: 133 Calculate the percentage of overall bandwidth saved (at Layer 2) by cRTP for a G. Approximately C. Approximately B. Approximately D.729 VoIP call packetized at 50 pps running over a MLP link. D. Approximately Answer: B Explanation: QUESTION NO: 134 Refer to the exhibit. Two ports on a 3600 gateway platform are stuck in the EM_PARK state. C. A. Approximately E. What workaround can be configured on the router to help alleviate this situation? A. B.

Same and Distinct C.cnf file instead of an SEP (mac address) file? (Choose 3) A. C. B.D Explanation: QUESTION NO: 136 Two type of RSVP reservation types are: A.Answer: A Explanation: QUESTION NO: 135 Based upon your understanding of an SCCP 7960 IP phone bootup sequence. The IP phone should not have previously registered with this CallManager cluster. Shared and non-shared D. D.cnf file explicitly. but the IP Phone was power cycled. E. what must be true for the CallManager to send the IP phone an SEPDefault. Auto Registration must be enabled. The dial-plan for the CME site is listed below.B. CompanyA: 4085551111 CompanyB: 4085552222 . Reservation and Path Answer: A Explanation: QUESTION NO: 137 A CME site has a group of attendants that answer calls for 3 different companies. Distinct and Shared B. The IP phone must request SEPDefault. Answer: A. The IP phone must be configured in the "default" partition. The IP phone must have previously registered.

..102 preference 1 timeout 5 B.. Incoming DNIS call to company A. The name of the company that was called should appear on the phone display.101.CompanyC: 4085553333 AttendantA: 100 AttendantB: 101 AttendantC: 102 Identify the configuration that will meet the following criteria: 1.....101. telephony-service directory entry 1 4085551111 name CompanyA directory entry 2 4085552222 name CompanyB directory entry 3 4085553333 name CompanyC service dnis dir-lookup ephone-hunt 1 longest-idle pilot 105 secondary 408555. A. list 100. 3. telephony-service directory entry 1 4085551111 name CompanyA directory entry 2 4085552222 name CompanyB directory entry 3 4085553333 name CompanyC service dnis dir-lookup ephone-hunt 1 longest-idle pilot 408555.. Attendants can logout so that calls to Company A. ephone-dn 1 number 100 secondary 408555. name CompanyA preference 1 no huntstop ! . B or C's dial-in number will hunt to the attendant that has been idle the longest. B or C will bypass their phone.. 2. list 100.102 preference 1 timeout 5 C.

.... name CompanyB preference 2 no huntstop ! ephone-dn 3 name CompanyA number 102 secondary 408555. name CompanyA preference 1 no huntstop ! ephone-dn 2 number 408555...ephone-dn 2 number 101 secondary 408555. telephony-service directory entry 1 4085551111 name CompanyA directory entry 2 4085552222 name CompanyB directory entry 3 4085553333 name CompanyC service dnis dir-lookup ephone-dn 1 number 100 secondary 4085551111 preference 1 no huntstop ! ephone-dn 2 number 101 secondary 4085552222 preference 2 no huntstop ! ephone-dn 3 number 102 secondary 4085553333 preference 2 no huntstop ....... preference 3 no huntstop E.. name CompanyB preference 2 no huntstop ! ephone-dn 3 name CompanyC number 408555. ephone-dn 1 number 408555... preference 2 no huntstop D.

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The 'OutputExternalFormat' parameter is not set correctly. To support remote sites for multi-site distributed call processing models and H. B. The 'External Phone Number Mask' for extension 2001 is not set correctly. Answer: C Explanation: QUESTION NO: 139 Consider the exhibit. A call placed from extension 2000 to extension 2001 is forwarded to voicemail and a message is recorded but MWI light is not turned on. D.323v1. To support remote sites in a centralized call processing model and H. The 'InputDnSignificantDigits' parameter is not set correctly.323v1.Answer: A Explanation: QUESTION NO: 138 A software media termination point should be deployed to support which call processing models and services? A. The 'OutputDnFormat' parameter is not set correctly.323v1. To support IP Telephony Service Providers and H. Answer: D Explanation: QUESTION NO: 140 Which would be situations where configuring and using a DNS server would be advisable in an IP . To support single site call processing models and H.323v1. what is the most likely cause of this problem? A. C. From the SMDI trace. B. D. C.

E. F. The MTU size in the network is longer than the bandwidth will allow.telephony network. G. If Network Address Translation (NAT) is required for communication between the IP phones and Cisco CallManager. B. Answer: A. On which two of the links (labeled A through F) would SCCP be used as the protocol? (Choose 2) A. D. A B C D E F . so only the first 1500 bytes are returned by the DNS server.E Explanation: QUESTION NO: 141 Consider the exhibit. DNS names resolution is required within the cluster deployed in a single site. IP telephony disaster recovery network configurations. DNS names resolution is required for Multi-Site WAN Deployments with Distributed Call Processing. (Choose 2) A. C. F. B. D. DNS names resolution is required for Multi-Site WAN Deployments with Centralized Call Processing. E. The size of the DNS response exceeds 1500 bytes. C.

A-Link or F-Link C. F-Link only Answer: B Explanation: QUESTION NO: 144 Given a traffic profile consisting of voice.Answer: B. Use Locations-based CAC with the branches and GK-based CAC with the Telephony Service Providers D. which class maps would correctly identify these applications? (Note: Assume the applications have already been previously . Use GK-based CAC with the Telephony Service Providers and with the branches Answer: C Explanation: QUESTION NO: 143 In Cisco's Thunderdial and Thundervoice solution using SS7. IMTs with Bearer channels E. is planning a deployment of Cisco IP Telephony using the Centralized Call Processing model. Use Locations-based CAC with the Telephony Service Providers and GK-based CAC with the branches B. T1 with IMTs D. T1/PRI B. interactive-video. Additionally. SLT can terminate which link type? A. transactional and bulk data. call signaling. a best-effort class and a "less-than Best Effort" scavengerclass. there will be an interconnection with multiple Telephony Service Providers using H323.F Explanation: QUESTION NO: 142 Company CADORNA's INC. Use Locations-based CAC with the Telephony Service Providers and with the branches C. The company has 1 HQ and 25 branches in a hub and spoke configuration. A-Link only F. Which Call Admission Control methods should be used in this network? A.

marked to their Cisco QoS Baseline default marking recommendations. Additionally. assume any .

application classes marked to an RFC 2597 AF PHB may have also been previously remarked by a policer. class-map VOICE match ip dscp 46 class-map INTERACTIVE-VIDEO match ip dscp 34 36 38 class-map CALL-SIGNALING match ip dscp 24 class-map TRANSACTIONAL-DATA match ip dscp 18 20 22 class-map BULK-DATA match ip dscp 10 12 14 class-map SCAVENGER match ip dscp 8 D. class-map VOICE match ip dscp 46 class-map INTERACTIVE-VIDEO match ip dscp 30 32 34 class-map CALL-SIGNALING match ip dscp 24 class-map TRANSACTIONAL-DATA match ip dscp 20 22 24 class-map BULK-DATA match ip dscp 10 12 14 class-map SCAVENGER match ip dscp 8 C. class-map VOICE match ip dscp 46 class-map INTERACTIVE-VIDEO match ip dscp 41 42 43 class-map CALL-SIGNALING match ip dscp 24 class-map TRANSACTIONAL-DATA match ip dscp 21 22 23 class-map BULK-DATA match ip dscp 11 12 13 class-map SCAVENGER match ip dscp 10 B. class-map VOICE match ip dscp 46 class-map INTERACTIVE-VIDEO match ip dscp 34 35 36 class-map CALL-SIGNALING match ip dscp 24 .) A.

C Explanation: QUESTION NO: 147 Bob's Bicycles is configuring Unity as an Auto Attendant.x server.37 Fax Over IP C.class-map TRANSACTIONAL-DATA match ip dscp 18 19 20 class-map BULK-DATA match ip dscp 10 11 12 class-map SCAVENGER match ip dscp 8 Answer: C Explanation: QUESTION NO: 145 In a CallManager 4. Fax Passthrough D. MGCP Answer: A. only ntp Answer: B Explanation: QUESTION NO: 146 Suppose your business has a need to send and receive real-time fax over a VoIP network. T. T. if the clocks on both systems need to be synchronized. ntpset B. The requirements are for outside callers . ntpclock D. which of the following commands will accomplish this task? A. timeset E.38 Fax-relay B. ntpdate C. Which fax protocol(s) could you implement? (Choose 2) A.

Which of the following is NOT a valid Skinny Call state for an IP Phone? A. 8 C. If they press 2 from the initial prompt they reach a submenu where they are prompted to "Press 1 to schedule an appointment or Press 2 to check the status of your repair". 11 Answer: B Explanation: QUESTION NO: 148 What is the relationship between a device pool and a region in CallManager configuration? A. At anytime the caller should be able to press 0 and be transferred to the Operator. 9 D. Press 2 for Accessories". Off Hook B. Press 3 for News and Events". D. Call Routing . Devices acquire a region setting from the device pool to which they are assigned. Devices use the settings in the region as the default pool for that region.to hear a prompt saying "Press 1 for Sales. 10 E. The device pool sets the default codec for the devices associated with it in a specific region and region sets only the codec that is used between other regions. If they Press 1. B. Press 2 for Service. The device pool sets the intra-region and inter-region codecs and the region makes the association between it and other regions. C. Answer: B Explanation: QUESTION NO: 149 In an Intra Cluster Call Flow Trace the following Skinny Call States could be observed. Unity should list the 4 sales people and allow the caller to choose. they reach a submenu where they are prompted to "Press 1 for Bike Sales. If they Press 1. Connected C. 7 B. What is the minimum number of Call Handlers required? A.

Call Waiting .D.

B.729 be used at all times. TTS D. Answer: B Explanation: Topic 4. ASR B. Database Answer: A. Which of the following statements is Not true? A. Volume D QUESTION NO: 151 What subsystems must be in service to enable speech recognition for the AA script? (Choose 2) A. Call Park Answer: C Explanation: QUESTION NO: 150 Company CADORNA's INC. The administrator will need to load-balance the load across the various CTI Managers in the cluster. It is recommended that all users be requested to accept a company note stating that 911 calls will not work properly due to the mobility nature of the Cisco IP Communicator. Call Transfer F. where a low-bandwidth codec such as g. is planning a deployment of Cisco IP Telephony using the Centralized Call Processing model. It is recommended that a new region be created for all the Cisco IP Communicators. JTAPI C. The company has 1 HQ and 25 branches in a hub and spoke configuration. The IT department will need to verify that the employees' laptop technical characteristics satisfy the Cisco IP Communicator's minimum requirements. RMCM E. D. The company wants to offer the option to use Cisco IP Communicator to some of its employees that are often on the road.E.B Explanation: . C.

Configure all intermediate devices to repeat the Progress Indicator signal E. regardless of which CallManager either device is registered to. the call will fail D. Ensure that the terminating gateway has not dropped the Progress Indicator signal D.) A. Configure the terminating gateway to send a PI=8 in the Alert message . Answer: B Explanation: QUESTION NO: 154 A POTS (PSTN/PBX) user places a call (through Cisco router/gateways) and does not hear a ringback tone before the call is answered.QUESTION NO: 152 What is the required bandwidth for 3 G. Which of the following actions would probably correct the situation? A. 192 kbps B. If the two phones are registered to different CallManagers and either CallManager fails.8 kbps Answer: C Explanation: QUESTION NO: 153 Which statement is TRUE? A. 240 kbps C. the call which is made on Inter cluster trunks is torn down.711 VoIP calls on an Ethernet link? (Note the packet per seconds count is set to 33 pps. B. 223. If two IP phones are registered to the same CallManager. 238. C. If the CM fails any call in progress is dropped. and the CM fails.5 kbps D. Ensure that all intermediate devices carry the correct Progress Indicator without modification C. the active call between those 2 phones goes down. If a call is made from an IP phone to a gateway (No VG2XX GW considered here). Configure the originating gateway to respond PI=1 to force ringback B.

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Adjusts volume levels of MOH source files D. D. Provides audio data from WAV files (ANN. Unity SMTP Networking C.F Explanation: QUESTION NO: 157 When deploying IPMA in proxy mode. Unity VPIM Networking Answer: A. Unity AMIS Networking D.Answer: E Explanation: QUESTION NO: 155 Which two of the following require specific Active Directory schema extensions? (Choose 2) A. B. Supports WMA and MP3. C. 3 3 2 2 partitions partitions partitions partitions and and and and 1 2 3 2 calling calling calling calling search search search search space spaces spaces spaces . MOH) E. what is the minimum number of partitions and calling search spaces required? A. Unity Bridge Networking B. Monitors a configured path for new files B. Converts new MOH source files to separate wav files for MOH codecs C.D Explanation: QUESTION NO: 156 Select 2 of the following which are NOT functions performed by MOH Audio Translator Service: A. Provides TFTP client download of MOH files Answer: D. F.

E. Performs digit manipulation C. Answer: A Explanation: QUESTION NO: 159 Which 2 functions are performed by a route group? A. The CTI Manager and Cisco TSP provide TAPI applications access to CallManager resources and functionality without being aware of a specific CallManager. Points to the actual gateway/gatekeeper Answer: B. B. 1 partition and 4 calling search spaces Answer: B Explanation: QUESTION NO: 158 Which function does CTI Manager provide for a CallManager cluster? A. the CTI Manager closes the provider and calls at JTAPI ports and route points that have not yet been terminated get redirected to the Call Forward On Failure (CFF) number that has been configured for them. Points to prioritized route groups F. Matches dialed number for external calls B. After a firewall is turned up. C. your PRI is no longer working. The CTI Manager and Cisco TSP provide both TAPI and JTAPI application access to specific CallManager servers in a cluster. Points to a route list for routing D.F Explanation: QUESTION NO: 160 You are implementing a CallManager solution utilizing MGCP gateways with PRI interface cards. When a Cisco CallManager node fails. Chooses path for call routing E. What is most likely the issue? . the CTI Manager recovers the affected JTAPI ports and route points. When a Cisco TAPI application fails. D.

the user has a user device profile that is configured for a Cisco IP Phone Model 7960 and logs into a Cisco IP Phone Model 7960). Both locations initiate video conferences. how does Cisco CallManager Extension Mobility behave? (Choose 2) A. The system uses the device profile default for that phone model for phone template and softkey template configuration and. for the addon module. CallManager TLS needs to be restarted after the firewall is installed to reestablish communication with the MGCP gateway. D. The user can access all of the services that are configured on their device profile. user hold audio hold audio source. B. speed dials. An L2TP tunnel needs to be set up between the CallManager and the firewall to carry signaling traffic Answer: B Explanation: QUESTION NO: 161 On authentication.E Explanation: QUESTION NO: 162 You have two locations with CallManagers. In each CallManager there are media resource groups: . Tampa and Atlanta. E. the services get copied from the user device profile. if the login profile matches the login device (that is. The phone automatically reconfigures with the individual user device profile information Answer: B. and directory number configuration except for the setting "line setting for this device") from the user device profile to the login device. The system copies all device-independent configurations (that is. user locale. B. The CallManager server needs to be on the same contiguous subnet as the gateway without a firewall between them. if the phone can support addon modules. The port range that the CallManager uses to communicate with the gateway is blocked. C. If the phone model supports Cisco IP Phone Services and they are configured. userid. D. C.A.

SW-CONF1 ATLHardware: HW-CONF2. XCODE1 TPASoftware: MTP1. VC-CONF1. VC-CONF2. XCODE2 . MOH1.TPAHardware: HW-CONF1.

C. B. In Tampa use the ATLHardware as primary and TPASoftware as secondary for video conferencing resources. A Unity server can service remote mailboxes if they are in the same Exchange 2000 Routing Group as the partner Exchange server. Unity should use the same Domain Controller as the Exchange servers.ATLSoftware: MTP2. In Atlanta use the ATLHardware as primary and ATLSoftware as secondary for video conferencing resources D. E. E. Answer: E Explanation: QUESTION NO: 163 Which three statements describe best practices when implementing Unity Unified Messaging in an Exchange 2000 environment? (Choose 3) A. MOH2. In Tampa use the TPAHardware as primary and ATLHardware as secondary for video conferencing resources. C. In Tampa and Atlanta use the local hardware resources as primary and the other sites hardware resources as secondary conferencing resources. In Tampa and Atlanta use the TPAHardware MRG as the source for video conferencing resources. Answer: A.D Explanation: QUESTION NO: 164 Identify the proper topology or the best way for deploying CAC in a centralized deployment: . B. Unity should be installed in the same Windows 2000 site as the Exchange servers. Each Unity server should have its own partner Exchange server. SW-CONF2 How should the Media Resource Group Lists be configured in the two locations so that video conferencing resources would be applied to video conferences? A. D. C. A Unity server should service a single Exchange Administrative Group.

A. Use H.323 Gatekeeper at a centralized location and use CallManagers at remote locations B. Use CallManager at a centralized location and use H.323 Gatekeeprs at remote locations C. Use single Cisco CallManager at a centralized location to control all of the remote locations D. Use Cisco CallManager in a cluster at a centralized location and also at remote locations Answer: C Explanation: QUESTION NO: 165 When using RSVP to dynamically set up end-to-end QoS across a heterogeneous network there are several QoS Levels that can be configured on the Originating and Terminating Gateway. Which of the following statements correctly describe the difference between these three QoS Levels: Controlled-Load, Guaranteed-delay and Best-effort? (Choose 3) A. With Controlled-load QoS Level; if synchronized RSVP is attempted and fails, the call is released. B. With Controlled-load QoS Level; if synchronized RSVP is attempted and fails, the call receives best-effort service. C. With Best-effort QoS Level; no RSVP synchronization is attempted and the call receives besteffort service. D. With Guaranteed-delay QoS Level; if synchronized RSVP is attempted and fails, if acceptable QoS on the terminating gateway is Controlled-Load or Guaranteed-delay, the call receives besteffort service. E. With Guaranteed-delay QoS Level; if synchronized RSVP is attempted and fails, if acceptable QoS on either gateway is anything other than best effort, the call is released. Answer: A,C,E Explanation: QUESTION NO: 166 An access-layer Cisco Catalyst 3550 is configured with voice and data VLANs. All QoS settings have been left at factory defaults, with two exceptions only:

Some access-edge ports are connected to Cisco IP Phones and other ports are directly connected to PCs. Assume that malicious users are setting the CoS and DSCP values of all their PCgenerated packets to CoS 6 and EF, respectively. Assume that the IP Phones are administered by

Voice: DSCP 40 Call-Signaling: DSCP Data: DSCP 0 B. Ensure that the IP Phone is set with the right time/date information D. Which would be LEAST helpful in solving the problem? A. When you pick up the handset or press the speaker button of the IP phone. Enure that the IP Phone shows at least an extension assigned to any buttons E. it can not make any outgoing calls. Voice: DSCP 40 Call-Signaling: DSCP Data: DSCP 48 D. Voice: DSCP 46 Call-Signaling: DSCP Data: DSCP 46 Answer: D Explanation: QUESTION NO: 167 An IP Phone appears to be registered to the Cisco CME system in "show ephone registered". Ensure that a button and ephone-dn association is shown in "show ephone register" Answer: C Explanation: 24 24 24 24 . Ensure that the IP Phone is registered to the right CME system by pressing Configuration -> Network Configuration -> TFTP server on the phone C. Voice: DSCP 46 Call-Signaling: DSCP Data: DSCP 0 C. you do not get a dial tone. Ensure that a 'button' command exists to associate a button with a configured ephone-dn in ephone configuration section. Ensure that the IP Phone is set to use the right CME system as the TFTP server B. However.Cisco CallManager 4. Add the 'button' command if needed F. as well as the (PC-generated) data packets as these packets exit the switch en route to the distribution layer? A. nor can it receive any incoming calls.0 (or higher). What will be the DSCP settings of the (IP Phone-generated) Voice and Call-Signaling packets.

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Each remote site has got its own voicemail solution using CUE(see site 2 for details). TAPI Service Provider (TSP) server component is only required on the partner Domino server.C. When 1001 calls 3001. One Unity server can service up to five Domino servers in a single Domino domain. Unity must be a domain controller or a member of a Windows domain.711 or configured the CCM in Headquarters to support G. but gets disconnected after a few rings. Check if there is a codec mismatch by either forcing the dial-peer on CME in site 2 to use G. Add a Transcoder on CME in Site 2 or in the network to convert to the right codec before transmission the media . B. or when 3001 is picked up to answer.711 for local B. however. On CCM.QUESTION NO: 168 Which three of the following statements regarding implementing Unity in an existing Domino environment are correct? (Choose 3) A.E Explanation: QUESTION NO: 169 Consider the Exhibit. Which of the following actions should be taken? (Choose 3) A. D. turn Media Termination Point (MTP) on D. On CCM. Site2 needs to be integrated with the Cisco CallManager deployed in headquarters to allow calls between CCM phones 3xxx at Headquarters and CME at remote sites (1xxx and 2xxx).729 for WAN and G. 1001 hears ring tones. C. Unity can be installed on the Domino server if there are less then 1000 mailboxes. Regarding CCM Integration with the CME system in Site 2: Site 2's phones still can not call phones in Headquarters. E. The Domino server that Unity communicates with must be in the same Windows domain as Unity. Answer: A. turn Media Termination Point (MTP) off C. ABC bank has added two remote sites (Site 2 and Site 3) with a Cisco CME running per each site.

Which of the following best describes how to configure the remote CallManager Express systems so calls can be forwarded to the central Unity server? A. The remote CallManager Express systems are configured with a VoIP dial peer that directs calls to the Unity pilot number to the central CallManager Express. D. The IP voice media streaming application needs to be started on the CallManager for the conference bridge to register.D Explanation: QUESTION NO: 170 A single Unity server is being deployed to provide voicemail for multiple CallManager Express systems.C. Each remote CallManager Express system is defined as a CallManager cluster in the Unity server. Which of the solutions will resolve this issue? A. B. The DSP conference resources on the IP gateway are improperly configured and will need to be deleted and reconfigured. One CallManager Express is colocated with the Unity server and the rest are connected via a WAN. Each remote CallManager Express is configured with a VoIP dial peer for each port assigned to it in Unity. Each remote CallManager Express is configured with ephones which register with dedicated ports in the Unity server.Answer: A. The software conference bridge requires external DSP resources to be configured for this service to operate correctly. When users try to use the conference bridge they get the error "No conference bridges available". The CallManager needs to be stopped and restarted for the conference bridge to register. D. C. B. C. Answer: D Explanation: QUESTION NO: 171 You have a CallManager with a software conference bridge. Answer: B Explanation: .

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Approximately Answer: E Explanation: QUESTION NO: 173 Which of the following will NOT make a good requirement for hardening an MCS Server's Operating System and its services? (Select 2) A. Harden accounts and passwords E. An intrusion protection system: CSA F. A. Approximately C. What are possible causes of this problem? (Choose 2) 60% 50% 40% 30% 20% . Approximately D. Keep the OS up to date and install Cisco OS upgrades that can be downloaded from Cisco Connection Online (CCO) B. Approximately B. Enable all services including file sharing D.711 VoIP call packetized at 50 pps. Ensure local applications' access to system registry Answer: C.QUESTION NO: 172 Calculate the percentage of overall bandwidth saved (at Layer 3) by cRTP for a G. While testing a new Cisco CallManager integration it is determined that IP phones can place and receive calls from PBX phones but cannot place calls to the PSTN. Approximately E. Harden the IP stack by mitigating redirection attacks and enabling SYN flood protection C.F Explanation: QUESTION NO: 174 Refer to the exhibit.

Authentication of callers B. The IP phones' CSS does not contain the partition assigned to the PBX users. B. The PBX is restricting trunk to trunk transfers.D Explanation: . Answer: A. Speech recognition and text to speech E. DTMF digit collection on VoIP dial peers Answer: B. The T1 to the PBX is using a different ISDN protocol than the T1 to the PSTN. D.D Explanation: QUESTION NO: 175 VXML and TCL provide similar services. Support for RTSP servers D. The CallManager is not sending the correct digits to the PBX. E. The gateway's CSS does not contain the partition assigned to the IP phones. C. Store and Forward of audio streams C.A. Which of the following are unique to VXML? (Choose 2) A.

as well as SRTP between gateway and IP phones? (Choose 3) A. CallManager 4. IPSEC tunneling in Windows 2000 of CallManager server E. IOS router configured in H323 D. IOS router configured in MGCP C. Protocol Call Manager Destination Port Skinny Gateway (Digital) TCP 2002 MGCP Control Message UDP 2427 SCCP TCP 2000 Skinny Gateway (Analogue) TCP 2001 RIS Data Collector TCP 2555 CTI TCP 2748 IPMA Service TCP 2912 C.D Explanation: QUESTION NO: 177 Identify the CallManager Destination Port that gets used with different Protocols as listed below: A.B. Protocol Call Manager Destination Port Skinny Gateway (Digital) TCP 2002 MGCP Control Messages UDP 2912 SCCP TCP 2000 Skinny Gateway (Analogue) TCP 2001 RIS Data Collector TCP 2748 CTI TCP 2555 .1 with CTL installed B. Protocol Call Manager Destination Port Skinny Gateway (Digital) TCP 2001 MGGP Control Messages UDP 2427 SCCP TCP 2000 Skinny Gateway (Analogue) TCP 2002 RIS Data Collector TCP 2555 CTI TCP 2912 IPMA Service TCP 2748 B.QUESTION NO: 176 Which of the following are the minimum requirements to have signaling encryption between CM and gateway. Router with NM-HDV module Answer: A.

IPMA Service TCP 2427 D. Protocol Call Manager Destination Port .

Master/Slave C. Peer-to-Peer B. G729 and Cisco Wideband conferences.X Software conference bridges have the following characteristics: (Choose 2) A.Skinny Gateway (Digital) TCP 2001 MGCP Control Messages UDP 2427 SCCP TCP 2002 Skinny Gateway (Analogue) TCP 2000 RIS Data Collector TCP 2555 CTI TCP 2748 IPMA Service TCP 2912 E. Protocol Call Manager Destination Port Skinny Gateway (Digital) TCP 2002 MGCP Control Messages UDP 2748 SCCP TCP 2000 Skinny Gateway (Analogue) TCP 2001 RIS Data Collector TCP 2555 CTI TCP 2427 IPMA Service TCP 2912 Answer: B Explanation: QUESTION NO: 178 Which three of the following attributes would correctly describe MGCP? (Choose 3) A. Intelligent Endpoints Answer: B.711. Centralized dial plan management F. Uses a Proxy Server to communicate with Cisco CallManager E. or 128 in a Meet Me conference are supported. B. A maximum of 64 participants in Ad Hoc conferences.C. Call preservation on GW failover from one CCM server to another D. . It supports G.E Explanation: QUESTION NO: 179 CallManager version 4.

Answer: B. Which statement about the shown configuration is the most accurate? A.C. The Change the 'class-map' to: Class-map match-any voice_and_citrix match dscp EF match dscp AF41 . Separate transcoder resources are needed if calls using G711 codecs want to join a software conference bridge. Likewise. All low bit-rate (LBR) calls must be transcoded prior to joining the conference call. D. Voice packets are getting marked by another device in the network as DSCP value EF before reaching Router1.C Explanation: QUESTION NO: 180 Consider the exhibit and configuration. the requirement for voice packets needs to be given the highest priority and citrix packets needs to be given second priority. Naturally. packets coming from Citrix server are marked as AF41.

In the above configuration. You should configure 'citrix' class as CBWFQ using 'bandwidth' since voice packets need to be given the highest priority. The configuration is incorrect because you can not have two 'priority' queue within the same policy. Each remote site has its own voicemail solution using CUE(see Site 2 for . voice packets will be given the highest priority. voice packets need to be given the highest priority. Thus. you should move the 'class voice' up in the order so that 'class citrix' will be pushed down.And then change the policy to: policy-map voice-qos class voice_and_citrix priority 800 bandwidth 500 class class-default fair-queue As this change in configuration matches packets with 'EF' first in the order under 'class-map voice_and_citrix' it will get the highest priority in the policy as it matches the priority queue. C. D. B. 'class citrix' is configured first in the order so it will be given the highest priority. The configuration is incorrect because. as per the requirement. Change the 'class-map' to: Class-map match-any voice_and_citrix match dscp EF match dscp AF41 And then change the policy to: policy-map voice-qos class voice_and_citrix priority 1300 class class-default fair-queue This change in configuration matches packets with 'EF' first in the order under 'class-map voice_and_citrix' it will get the highest priority. Packets with AF41 will go into bandwidth queue since it is second in the list. Hence. ABC bank has added two remote sites (site 2 and site 3) with a Cisco CME running per each site. Answer: B Explanation: QUESTION NO: 181 Consider the exhibit.

What additional configuration commands are needed for CME/CUE integration? .details). Site 2 needs to be integrated with the Cisco CallManager deployed in headquarters to allow calls between CCM phones 3xxx at Headquarters and CME at remote sites (1xxx and 2xxx). however.

1.A.255.1.1.1 dial-peer voice 5777 voip destination-pattern 57777 session target ipv4:10.1.2 255. Configure the following on CME system to route the calls to the CUE module and voicemail .1.2 255.2 codec g711ulaw no vad C. Configure the following on CME system to route the calls to the CUE module and voicemail pilot#57777: Ip route 10.255 10.255.2 codec g711ulaw no vad B. Configure the following on CME system to route the calls to the CUE module and voicemail pilot#57777: Ip route 10.1.1.1.255.1.1.255 Service-Engine3/0 dial-peer voice 5777 voip destination-pattern 57777 session target ipv4:10.

C.1. Answer: C Explanation: .1.1.255.255 Service-Engine3/0 dial-peer voice 57777 voip destination-pattern 57777 session protocol sipv2 session target ipv4:10.1. B.2 dtmf-relay sip-notify codec g711ulaw no vad Answer: D Explanation: QUESTION NO: 182 Which of the following best describes how MWI is accomplished when integrating Unity with multiple CallManager Express systems? A.255. One CallManager Express system is configured as a SIP MWI server and relays MWI messages to the other CallManager Express systems. The appropriate CallManager Express system processes the MWI and the other CallManager Express systems discard the MWI. The Unity server sends MWI to the appropriate CallManager Express systems based on the cluster definition. D.255.1.2 255.1.pilot#57777: Ip route 10. Configure the following on CME system to route the calls to the CUE module and voicemail pilot#57777: ip route 10.2 255.255 10. Each CallManager Express system is defined as a CallManager cluster in the Unity server.1.2 Dtmf-relay sip-notify codec g711ulaw no vad D.1. One CallManager Express system is configured as a CallManager in the Unity server and relays MWI messages to the other CallManager Express systems using SCCP.1. The Unity server sends MWI to all CallManager Express systems.1.1 Dial-peer voice 57777 voip Destination-pattern 57777 Session protocol sipv2 Session target ipv4:10.255.

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one for the internal codec type and one for the IP WAN codec. Which two tasks need to be completed to support locations for this solution? (Choose 2) A. such as an IP phone Answer: B Explanation: QUESTION NO: 185 In a centralized call processing environment there are three sites. CallManager B. D. To ensure that a new voice flows on a statistically multiplexed circuit. Each site will use G. One region will be required for each site.729 between sites. C. call does not degrade existing voice call does not degrade existing voice call does not degrade existing data call does not degrade existing data . Signaling Link Terminal D. B. Answer: B Explanation: QUESTION NO: 184 When implementing PRI backhaul for an MGCP gateway and CallManager. Site A and Site B that connect over an IP WAN. the Q.921 data-link protocol is terminated on what device? A. The IP end device. Each site will require a region that assigns all devices to a device pool. Each site will require a device pool that specifies the site name in the region. To ensure that a new voice calls on a time division multiplexed circuit.QUESTION NO: 183 What is the purpose of CAC on a converged network? A. To ensure that a new voice flows on a time division multiplexed circuit. B. To ensure that a new voice calls on a statistically multiplexed circuit. MGCP gateway C.711 internally and G. HQ. C. D. Two regions will need to be configured for each site.

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Answer: B.C Explanation: QUESTION NO: 186 Which of the following are H. All sites will become a member of a single region with specific device pools for each site. Cisco IP Manager Assistant Answer: A Explanation: QUESTION NO: 188 An IPCC script is configured with a DB Read to check to see if it is a holiday. Which trace should be enabled to troubleshoot this issue? A. H. that is NOT a valid CallManager service parameter: A. Cisco RTMT Data Collector B. CODEC negotiations between calling and called party B. IPaddress exchange and UDP port negotiations between calling and called party Answer: A.E.245 standard functions? (Choose 2) A. Calls to the database fail intermittently. IP port negotiations between calling and called party D. Cisco Telephony Call Dispatcher E. Cisco Database layer Monitor C. Database Layer Monitor . Cisco Webdialer F.E Explanation: QUESTION NO: 187 Choose which one of these. Cisco Serviceability Reporter D. IP phone on-hook and off-hook signal exchange E.225 UDP port call setup negotiations between calling and called party C.

SS_DB Answer: D Explanation: QUESTION NO: 189 Suppose the following command is configured in a gatekeeper: "bandwidth total default 64'.729 codec. Enable RFC 2597 congestion avoidance techniques only on application classes assigned an AF PHProvision a "less than Best Effort" queuing policy for scavenger traffic (assume a non-distributed router platform). bulk data. RIS Data Collector D.B. CTI Manager C. transactional data. policy-map WAN-EDGE-QUEUING class VOICE priority percent 33 ip rtp header-compression class CALL-SIGNALING bandwidth percent 5 random-detect dscp-based class TRANSACTIONAL-DATA bandwidth percent 26 random-detect dscp-based class BULK-DATA . Which statement would be TRUE? A. Also assume that these traffic classes have been correctly classified and marked according to Cisco's best practices. This gatekeeper will admit up to 4 calls using G. regardless of the codec used. B. Enable class-based cRTP on the voice class.729 codec. best effort and a "less than Best Effort" scavenger traffic class. callsignaling. This gatekeeper will not admit any calls because all calls initially account of 128Kbps. This gatekeeper will admit a minimum of 4 calls using G. This gatekeeper will admit up to 64 calls. Answer: D Explanation: QUESTION NO: 190 Design a WAN edge LLQ/CBWFQ policy to accommodate voice. D. Which option meets this criteria? A. C.

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policy-map WAN-EDGE-QUEUING class VOICE priority percent 33 ip rtp header-compression class CALL-SIGNALING bandwidth percent 5 dscp-based random-detect class TRANSACTIONAL-DATA bandwidth percent 26 dscp-based random-detect class BULK-DATA bandwidth percent 10 dscp-based random-detect class SCAVENGER bandwidth percent 1 class class-default fair-queue D. policy-map WAN-EDGE-QUEUING class VOICE priority percent 33 compression header ip rtp class CALL-SIGNALING bandwidth percent 5 class TRANSACTIONAL-DATA bandwidth percent 26 random-detect dscp-based class BULK-DATA bandwidth percent 10 random-detect dscp-based class SCAVENGER bandwidth percent 1 class class-default bandwidth percent 25 C.bandwidth percent 10 random-detect dscp-based class SCAVENGER bandwidth percent 1 class class-default faor-queue random-detect. policy-map WAN-EDGE-QUEUING class VOICE priority percent 33 ip rtp header-compression class CALL-SIGNALING bandwidth percent 5 dscp-based random-detect . dscp-based B.

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which contains 32 bits for the tag protocol ID.1Q header is included.1Q header is included. 802. 3 bits for the priority field. In Ethernet frame after the source address. 6 bytes 802. Size of 802. Size of 802.class TRANSACTIONAL-DATA bandwidth percent 26 dscp-based random-detect class BULK-DATA bandwidth percent 10 dscp-based random-detect class SCAVENGER bandwidth percent 1 dscp-based random-detect class class-default fair-queue random-detect dscp-based Answer: B Explanation: QUESTION NO: 191 Which of the following is true about the 802.1q frame header? A.1q frame is inserted inside the actual Ethernet frame. 1 bit for the canonical field (always 0). PCI voice cards are not required for the Unity server. 3 bits for the priority field.1q frame is 4 bytes. 4 bytes 802. it is inserted after Source Address in the Ethernet frame. C. B. Answer: B Explanation: QUESTION NO: 192 A Unity server is being deployed in a dual integration with CallManager and a PBX. The CallManager integration must be completed first.1q frame is 26 bytes. In an Ethernet frame after the source address. C. The PBX integration will utilize a PBXLink. D. B. In other words. 1 bit for the canonical field (always 0).1q frame is inserted at the beginning of Ethernet frame. Which two of the following statements are true? (Choose 2) A. 802. which contains 16 bits for the tag protocol ID. . and 12 bits for the VLAN identifier. and 12 bits for the VLAN identifier. Each PBXLink port will connect to an analog port in the PBX.

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The next call will not use an MTP and will use the transcoding resources associated with the codec to complete the call. and an n + 1 connection is attempted how will the next call be treated? A. The PBXLink will communicate with Unity using SMDI. E. which of the following must be associated with the JTAPI user? (Choose 2) A. C. D.E Explanation: QUESTION NO: 193 If all n MTP transcoding sessions are utilized. The PBX and the PBXLink must be co-located. CSS defines search for directory numbers in assigned partitions according to dial patterns. The next call will be redirected to the PSTN due to a lack of MTP resources. The next call will complete without using the MTP transcoding resource Answer: D Explanation: QUESTION NO: 194 Which of the following does NOT describe the Calling Search Space function in CM properly? A. The Calling Search Space defines what numbers are available to a device to call. The next call will use the alternate codec type and attempt to complete the call D. Answer: D. C. E. The CSS defines Route Patterns and Directory Numbers calls can be received from. Calling Search Space is a grouping of partitions to look through when making a call. B. Each directory number has CSS.D. Agent devices . Answer: D Explanation: QUESTION NO: 195 When configuring ICD. B.

The company has 1 HQ and 25 branches in a hub and spoke configuration. Use MoH directly generated from the IP Phones in the Branches Answer: A.B. Use Multicast flows in the HQ and Unicast flows in the branches C. The company policies determine that multicast traffic should not be carried over the WAN. Use Multicast flows throughout the network.X and IP Phones ? (Choose 2) A. Which of the following statements would not violate the company policies while successfully implementing Music on Hold throughout all the different sites? (Choose 3) A. Agent ICD numbers C. Use Unicast MoH throughout the network D. If EM user has more than one device profile associated with him. Hunt Pilot Answer: C. E. A device profile needs to be created and assigned with Extension Mobility user in the CallManager B.C Explanation: QUESTION NO: 197 Which of the following steps will not fit into Extension Mobility(EM) service login or logout flow on Cisco CallManager 4.D Explanation: QUESTION NO: 196 Company CADORNA's INC. CTI Ports D. The EM user is associated with one of the device profiles. he is prompted to select the profile he wants to log in with. is planning a deployment of Cisco IP Telephony using the Centralized Call Processing model. Extension Mobility IP Phone Service URL is created for both Extension Mobility login and for Extension Mobility logout.B. filtering Multicast flows in the WAN edge. C. and stream MoH streams from the flash of the branches' routers B. CTI Route Points E. Use the AMOHB (Advanced Music on Hold for Branches) feature in Cisco CallManager E. The EM user's profile is loaded on the phone without the phone . The EM user selects the EM Service on the IP phone. D. F.

breaking its TCP connection with CallManager. .

Answer: B Explanation: QUESTION NO: 200 Which of the following does NOT accurately describe the Device Pool function in CM? A. D. iDivert D. pool allows Failover preferences to a phone or gateway. A VIP 30 IP phone is configured for each call path. .F Explanation: QUESTION NO: 198 Which of the following CallManager 4. ConfList Answer: C Explanation: QUESTION NO: 199 A DPA 7630 is being used to integrate a CallManager cluster with an Octel voicemail system. E. The Device gateway. C. A CTI port is configured for each call path. CallBack B. C. pool allows SRST Reference to a phone or gateway.Answer: B. The Device B. The Device Pool allows to add Call Manager group to a device. Pool allows Registration preferences to a phone or pool allows Location preferences to a phone or gateway. CTI Route Point C. A voicemail port is added for each call path. The Device E. A CTI route point is added with a DN for each call path. A 7902 IP phone is configured for each call path. The Device D. What is configured in the CallManager cluster to establish call paths to the DPA 7630? A.1(2) Applications require JTAPI support? A. B.

. link 6.C Explanation: QUESTION NO: 202 Consider the exhibit. How can the transfer issue be resolved? (Choose 2) A. ' D. . Link 2. Volume E QUESTION NO: 201 IP phones A (extension 1001). B. Link 8 Link 1. D. B (extension 1002). Link 6. . . when any of the phones attempt to transfer a call to any of the other phones they get a fast busy tone. Link 5. Link 8 Link 1. Link 3. C. C. Link 3. and C (extension 1003) are registered to the CME system and are able to call each other by dialing each others' extension numbers. (Please note that the Call Manager version is 4. please identify all the links between the devices shown where SCCP (Skinny Client Control Protocol) can be used for the two listed devices to communicate. Link 7 Link 1. However.x or below. or transfer-pattern . Link 2. Add this command under "telephony-service" on the CME system: transfer-system full-consult Answer: B.) A. Link 8 .Answer: C Explanation: Topic 5. Upgrade the IP phone firmware with the latest version. According to the list. Add this command under "telephony service" on the CME system: transfer-pattern 1 . Link 7. B. Reset the phonesso that the transfer capability will work. Link1. Link 2. Link 5.

E. All Links shown can use SCCP for the devices to communicate Answer: D Explanation: QUESTION NO: 203 Which ports must be opened on an IOS firewall to allow successful H.225 RAS (Registration, Admission, and Status) message exchanges between an IOS gatekeeper and IOS H.323 gateways? A. TCP 1718 and B. UDP 1718 and C. UDP 1718 and D. UDP 1719 and E. TCP 1718 and Answer: C Explanation: QUESTION NO: 204 Company CADORNA's INC. is planning a deployment of Cisco Contact Center Express and are looking for a design that provides maximum resilience, performance, and redundancy. They are planning to install Cisco CallManager 4.1(3) and Cisco Contact Center Express 4.0. How many CTI Managers can we configure in one cluster? A. 1 B. 2 C. 4 D. 8 E. 16 Answer: D Explanation: QUESTION NO: 205 Company CADORNA's INC. is planning a deployment of Cisco IP Telephony using the Centralized Call Processing model, using Cisco CallManager 4.1(3). The company has 1 HQ and TCP TCP UDP UDP TCP 1719 1719 1719 1720 1720

25 branches interconnected with an MPLS network that provides "fullmesh" connectivity between all sites. Which of the following statements is NOT true? A. This topology implies that, from an IP routing perspective on the enterprise side of the network, each site is one IP hop away from all other sites. B. From the Cisco CallManager's call admission control perspective, a service-provider IP WAN service based on MPLS is in reality equivalent to a hub-and-spoke topology without a hub site. C. This configuration requires that call admission control be performed on the central site link independently of the branch links. D. The Cisco CallManager Servers need to be aware of the underlying MPLS network by setting the appropriate Enterprise Parameters. E. All of the above statements are true. Answer: D Explanation: QUESTION NO: 206 If the CFNA and CFB are set to a hunt pilot number, what impact will the maximum hunt timer have on a call sent to the hunt pilot? A. The timer limits the number of seconds allotted for hunting through a hunt list. B. The timer limits the number of seconds the call will wait for an answer at each member of a line group. C. The timer limits the number of seconds a call will wait before being forwarded to a voicemessaging system, a specific dialed number, or some personal treatment (if configured), or the call gets released. D. The timer allows limits on the number of seconds allotted to hunting through a line group. Answer: A Explanation: QUESTION NO: 207 Two ports on a 3600 gateway platform are stuck in the EM_PARK state. What are two possible causes of this problem? (Choose 2) A. A call is parked and no one has answered.

B. A fake answer has been configured on the router. C. The DSP is having hardware or software issues.

For DHCP snooping to function properly. Which of the following . DHCP Snooping stops Man-in-the-Middle (MITM) Attacks Answer: E Explanation: QUESTION NO: 209 Acme Widgets Inc. 16. all DHCP servers must be connected to the switch through trusted interfaces. 12.D. MTP is required for call termination and for becoming RFC 2833 compliant.D Explanation: QUESTION NO: 208 Which of the following is NOT true about DHCP snooping? A. 8.0 kbps Answer: C Explanation: QUESTION NO: 210 For a phone call going out of Call Manager on SIP trunks to SIP endpoints in CM 4.0 kbps E. 8. The PSTN switch / PBX is sending a continuous off-hook signal Answer: C. DHCP snooping is a feature that provides network security by filtering untrusted DHCP messages and by building and maintaining a DHCP snooping binding database. DHCP Snooping is enabled by VLAN.729 loading two voice frames per packet. 9. C. wants to compress the voice data traveling over their WAN connection to their parent company.X version.6 kbps D. D. what will be the bandwidth consumption per call over the HDLC WAN link? A. When they implement cRTP using the ip rtp header-compression command. E. B.0 kbps B. DHCP Snooping has the capability to use rate limiting.8 kbps C. They are presently using the G.

Calls to the PSTN are being rejected. E. D.is Not a correct explanation for using MTP for this call? A. This payload type is fixed and not negotiated between CallManager and SIP endpoints via the SIP messages and is passed to MTP during media establishment. This payload type is configurable in Service Parameter as "SIPDefaultTelephonyEventPayloadType". RFC 2833 defines a dynamic payload type for DTMF tones. Change the ISDN Numbering Plan type to International Answer: D Explanation: QUESTION NO: 212 . Default value as 101. C. Which of the following will correct this issue? A. Standard for SIP DTMF is based on RFC2833 which uses in-band payload types to indicate tones. Add the command 'isdn contiguous-bchan' to the serial interface C. DTMF relay between in-band and out-ofband digits are accomplished. B. Add the command 'isdn negotiate-bchan' to the serial interface B. Change the channel selection order to ascending D. Answer: D Explanation: QUESTION NO: 211 Refer to the shown output. By making SIP calls use MTP.

MSDP. 4-d . . CGMP . 5 . 2 -b Answer: C Explanation: QUESTION NO: 213 Which of the following does not accurately describe the database layer service in CM? A. . 2 -b B. For real-time processing and call processing. 1 .e.a 3-a. B. The publisher maintains a TCP connection with each subscriber database. 1 . All remaining machines in a cluster are subscriber databases. 2 -d D. and modification of the database. 4-e . . 3-c.e. Auto RP. 1 . 5 . 2 -c C. RGMP. Anycast RP. 1) 2) 3) 4) 5) a) b) c) d) e) Basic Multicast Reliable Multicast One-to-Many Multicast Inter-Domain Multicast Many-to-Many Multicast MBGP . BSR SSM and IGMP v3 PIM . The database information itself is made up of one publisher database. 5 . 5 . IGMP v2.a. The database layer is a set of DLLs that provide a common access point for data insertion. each CallManager maintains a TCP connection with every other .e A.b 3-b. C.Bi-Directional PGM PIM SM.c 3-c . retrieval. 4-a .d. DM. which is the very 1st CallManager Machine in the CM cluster. 1 . 4-d .Match the IP Multicast Components as per their simple definitions.

E Explanation: . 20 ms Sample. 64 bytes B. Performs digit manipulation C. What would be data frame size coming out of the phone as seen in the Sniffer? A. 204 bytes F. Answer: E Explanation: QUESTION NO: 214 An IP Phone configured in voice VLAN 128 is registered to a CM cluster. Points to prioritized route groups F. Points to a route list for routing D. 218 bytes Answer: F Explanation: QUESTION NO: 215 Which 2 functions are performed by a Route list? A. it will try a replicated database for device information and monitor the publisher for its return. Chooses path for call routing E. Points to the actual devices Answer: D. 200 bytes E. Matches dialed number for external calls B. It has speech connection established with another IP Phone which is connected to another CM cluster. 128 bytes C. D. E. A PC is connected to back of an IP Phone which is running Sniffer Program and collecting packets coming out of the IP Phone. The database layer monitors access to the Publisher.711. All database records including Call Detail Records (CDR) are replicated from publisher to subscriber.CallManager via the Intra Cluster Communications Signalling (ICCS) protocol. Configured codec on both phones is G. 160 bytes D. During a failure.

D. An IP multicast stream to the IP host can only be stopped by an IGMP Leave message. The CallManager administrator uses the Cisco IP Telephony Services Configuration menu to develop separate lists of users and services.E Explanation: QUESTION NO: 217 Which of the following is NOT true about Multicast IGMP Snooping? A. The CallManager administrator can add services to Cisco IP phones and device profiles.QUESTION NO: 216 Which two ways are customized phone services subscribed to? (Choose 2) A. B. F. E. IGMP snooping requires the LAN switch to examine. Answer: C. The CallManager administrator uses the Cisco IP Phone Services Configuration menu to define and maintain the list of Cisco IP Phone services to which users can subscribe at their site. D. When the switch hears the IGMP host report from a host for a particular multicast group. Users can log into the Cisco CallManager User Options Menu and select the services they want to subscribe to and CallManager will automatically configure those services. they are indistinguishable from multicast data at Layer 2. some Layer 3 information in the IGMP packets sent between the hosts and the router. A switch running IGMP snooping examine every multicast data packet to check whether it contains any pertinent IGMP must control information. the switch adds the host's port number to the associated multicast table entry. C. Users can log into the Cisco CallManager User Options Menu to subscribe to services already configured by the CallManager administrator. Answer: B Explanation: QUESTION NO: 218 . C. B. only that port is deleted from the multicast group. or snoop. Because IGMP control messages are transmitted as multicast packets. E. When a host in a multicast group sends a IGMP leave message.

Why has Cisco chosen to use the SCCP protocol in its IP telephony networks? .

DN and Phone Model of phones that are registered/while registering? (Choose 2) A.A. Configure MAC address from Ethernet interface on the PC where IP communicator will be installed in Phone Configuration screen in CallManager Answer: A. Verify CallManager version B. B.E Explanation: QUESTION NO: 221 Given a Catalyst 3550 configured to support IPT. open protocol.E Explanation: QUESTION NO: 220 In CME. show ephone D. It uses intelligent endpoints. Configure a new CTI port to connect IP Communicator to CallManager D. IP Address. debug ephone register Answer: C. Associate directory number from desk phone with IP Communicator software E. show ephone-dn C. which configuration would correctly identify voice . which two steps need to be accomplished in order for the application to operate correctly? (Choose 2) A. It is a peer to peer protocol. Answer: D Explanation: QUESTION NO: 219 When implementing IP Communicator in CallManager. debug ephone detail B. which command/debug will show MAC Address. It is an industry standard. show ephone-dn registered E. Create and associate a user ID with the communicator device C. C. D. It enables the use of a rich set of features.

The Voice Mail Box Mask in CallManager is set to 40000 instead of XXXXX. class-map match-all VOICE match dscp 5 class-map match-all CALL-SIGNALING match dscp 3 D. Extension 75000 has not been defined in Unity.are set according to current Cisco best-practice recommendations. A site is using 5 digit extensions for internal calling. B. C.) A. What is the probable cause? A. A greeting has not been recorded for mailbox 75000. Calls to extension 75000 hear the Unity Opening Greeting instead of the subscriber's greeting when forwarded to Unity. D. The voicemail pilot number is 40000. class-map match-all VOICE match ip dscp 46 class-map match-all CALL-SIGNALING match ip dscp 24 Answer: D Explanation: QUESTION NO: 222 Consider the exhibit.and call signaling traffic originating from a Cisco IP Phone? (Note: Assume that all other QoS configurations .both on the switch and on the Cisco CallManager . class-map match-all VOICE match dscp ef class-map match-all CALL-SIGNALING match dscp af31 B. . class-map match-all VOICE match cos 5 class-map match-all CALL-SIGNALING match cos 3 C. The call CUGA routing rule Attempt Forward to Greeting does not exist.

How should the MoH services be configured for this company? A. The CallManager integration must be completed first. C.D Explanation: . Which two of the following statements are true? (Choose 2) A. Use multicast to stream the MoH files for the sales departments and multicast for a single stream for administration. The PBX integration will utilize a PIMG. Use unicast to stream the MoH files for the sales departments and administration. Use multicast to stream a different message to each sales department caller and unicast to each caller for each administrative group. B. Each department in the company needs to have a specific MoH message for their callers. Use unicast to stream the MoH files for the sales departments and multicast for a single stream for administration. There are 12 departments in the company. Use multicast to stream one message to each caller for all the sales departments and unicast to stream a different set of messages to each caller for each administrative group. PCI voice cards are not required for the Unity server. The PBX and the PIMG must be co-located. E. B. The 12th department is administration and will require specific messages for each group within that department. The PBX will require an analog port for each PIMG port.Answer: A Explanation: QUESTION NO: 223 A company would like to provide marketing messages when callers are placed on hold. Answer: E Explanation: QUESTION NO: 224 A Unity server is being deployed in a dual integration with CallManager and a PBX. D. C. The PIMG will communicate with Unity using SCCP. E. D. Answer: B. Eleven of the departments are sales oriented and need to market to customers with MOH.

QUESTION NO: 225

ACME is migrating from Exchange 5.5 to Exchange 2000. They have setup an Exchange MixedMode Environment. Which of the following statements regarding integrating Unity into this environment are correct? A. The partner server must be an Exchange 2000 server for Unity to service both Exchange 5.5 and Exchange 2000 mailboxes. B. One Unity server partners with an Exchange 2000 server and one Unity server partners with an Exchange 5.5 server. Digital networking is used to forward messages between the systems. C. The Unity server can support multiple Exchange 5.5 organizations but only one Exchange 2000 domain. D. The Unity server must be voice-mail only until the Exchange migration is complete. Answer: A Explanation: QUESTION NO: 226 Which two of the following attributes would correctly describe H.323? (Choose 2) A. Peer-to-Peer B. Master/Slave C. Uses a Proxy Server to communicate with Cisco CallManager D. Centralized dial plan management E. Intelligent Endpoints Answer: A,E Explanation: QUESTION NO: 227 There are customer complaints about voice quality between IP Phones. What could be the potential problem? (Assume that all switchports are configured to trust DSCP and that the WAN link is less than 768 kbps).

A. It is NOT possible to configure multiple DSCP matches in one statement under 'class-map'. So, the packets that are marked as DSCP=5 by IP phones are not going through priority queue. Instead, it is going through 'default queue '. So, IP phone packets are getting delayed. Thus, it caused the voice quality problem. B. It is possible to configure only TWO DSCP matches in one statement under 'class-map'. Due to this invalid config the packets that are marked as DSCP=5 by IP phones are not going through priority queue. Instead, it is going through 'default queue '. So, IP phone packets are getting delayed. Thus, it caused the voice quality problem. C. It is possible to configure multiple DSCP matches in one statement under a given class-map. Since the "voice-qos" class is matching three different DSCP values, they are all being assigned to the same "priority" queue, where they are processed in a FIFO manner, which is causing excessive delays on the slow-speed WAN link. D. None of the above. Answer: C Explanation: QUESTION NO: 228 Unity extensions will increase the Active Directory size by approximately what percent? A. 5 B. 10 C. 15 D. 20 Answer: B

. @ QUESTION NO: 230 Which R2 signaling element passes address information such as callingand called-party numbers? A. Cisco Unified Contact Center is not able to resolve the host name of Cisco Unified Communications Manager. 123@ B. line signaling D. 123? Answer: D Explanation: The valid route pattern caharacters are [ ^ 0 1 2 3 4 5 6 7 8 9 .] + ? ! X * # + . delay dial signaling C. A referenced CTI Route Point is not associated with the JTAPI user. 123$ E. interregister signaling E. out-of-band signaling Answer: E Explanation: QUESTION NO: 231 Which three of these are possible reasons why a JTAPI subsystem might have the status PARTIAL_SERVICES? (Choose two) A. pulse signaling B. C. 123* D. B. 123.QUESTION NO: 229 Which string is not a valid route pattern on Cisco Unified Communication Manager? A.

G. G.711. G. G 729 F.726 E. D. G.CNG algorithm. Answer: B.711 mu-law B.727.728 etc QUESTION NO: 233 Refer to the exhibit.F Explanation: This VAD-CNG mechanism can be used with any speech codec.C.C Explanation: QUESTION NO: 232 Which two codecs provide built in VAD? (Choose two) A. G722-64K C. G. G 729 Annex B Answer: C. The JTAPI user password is not correct. G. . like ITU-T G. without built in VAD.723.1 Annex A D.726. There is an error in one of the scripts being loaded.

5 D.What is the maximum number of inbound calls to 2001 before a Cisco Unified Communications Manager Express system returns user busy tone to any additional calls? A.323 Fast Connect (also known as fast start) procedure are not correct? (Choose four) . 7 Answer: E Explanation: QUESTION NO: 234 Which statements about the H. 4 C. 3 B. 6 E.

225 call setup and call connect messages. E. F.D. time slots C. Fast Connect allows logical channel negotiations earlier in the H. time slots B. .225 stage. Answer: B. B. Fast Connect Open Logical Channel elements can be carried in H.323 endpoint. time slots Answer: B Explanation: 0 0 1 0 to to to to 31 15.A. Fast Connect was introduced in H. and 17 to 32 QUESTION NO: 236 Refer to the exhibit. time slots D. Fast Connect was introduced in H.323 version 2.F Explanation: QUESTION NO: 235 Which E1 time slots are used to carry encoded voice only? A.C.225 facility messages. and 17 to 31 32 15. The Fast Connect proposal can be refused by the receiving H. D.323 version 3. C. Fast Connect Open Logical Channel elements can only be carried in H.

dial-peer voice E. dial-peer voice D. dial-peer voice C. 2 3 5 8 . None of above Answer: B Explanation: QUESTION NO: 237 Refer to the exhibit.Which dial peer will the Cisco IOS voice gateway match if an incoming call with a called number of 100 arrives at this T1 PR1 port? A. dial-peer voice B.

intelligent endpoints Answer: A. A. Master/Slave C. B C. is used as TPID? A. D Answer: A Explanation: QUESTION NO: 238 Which three attributes correctly describe aspects of SIP? (Choose three) A. C and D. peer-to-peer B.D. call preservation on gateway failover from one Unified Communications Manager server to another D. which block of bits. B.In this 802. B. communication with Unified Communications Manager handled via a proxy server E. C D. centralized dial plan management F. labeled A. .F Explanation: QUESTION NO: 239 Refer to the exhibit.1Q tagged Ethernet frame.

Cisco SSCA and email.cisco.com”? A. Cisco SSCA only C.cisco.Which of the certificates that are shown must be uploaded to Cisco Unified Presence when integrating the calendar with Exchange Server “email. gateway C. email.com Answer: C Explanation: QUESTION NO: 240 When using Local Route Group feature in Cisco Unified Communications Manager. in which two levels can you apply the called party transformation pattern? (Choose two) A. route pattern . DST Root CA X3 only B. DST Root CA X3 and Cisco SSCA E. device pool B.cisco.com D.

SIP B. which two protocols satisfy the following three requirements? (Choose two) Requirement 1: the protocol has a mechanism for a centralized dialplan Requirement 2: the endpoint are considered to be unintelligent Requirement 3: the protocol is text-based A. route list F. SSCP Answer: C. H. 13 C. 0 B. where default mappings between RTP payload type numbers and encodings are defined. MGCP D. route group E.D. which RTP payload type corresponds to encoded packets that are triggered by silence on a call with voice activity detection? A.323 C. service parameter Answer: C. 15 .D Explanation: QUESTION NO: 241 In a VoIP deployment.D Explanation: QUESTION NO: 242 According to RFC 3551.

18 Answer: B Explanation: http://www. Which of these does not belong to the supplementary services? A. router (conf-serv-sip)# transport tcp E. Call Back B. router (conf-voi-serv)#no sip transport udp C. router (conf-serv-sip)# no sip transport udp D. Message Waiting indicator D. router (config)#sip transport tcp B. router (config-sip-us)# no transport udp Answer: D Explanation: . Path Replacement Answer: B Explanation: QUESTION NO: 244 Which Cisco IOS Command and configuration mode can be used to force a Cisco IOS voice gateway to use TCP as the transport for SIP? A. while payload type 18 corresponds to codec G729 QUESTION NO: 243 When calls are placed by certain Cisco Unified Communications Manager supplementary services.org/assignments/rtp-parametersAs this link shows payload type 13 corresponds to silence suppression. Mobility Follow Me E.iana. the Local Route Group feature will be bypassed.D. Call Forward C.

C. Cisco Unity Connection will automatically purge all deleted messages in the user mailbox. The user hears a warning that the message cannot be sent. Answer: B Explanation: QUESTION NO: 246 Which of these best describes packetization delay in a VoIP network? A. user the DSP to compress a block of PCM the compression algorithm to correctly fill a packet payload with to clock a voice frame onto the network queue a voice frame for transmission on . the time that is taken to the network connection Answer: C Explanation: QUESTION NO: 247 Cisco Unity extents a number of schema object classes in Microsoft Active Directory during the schema extension process. E. B. the time that is taken by process samples block N C. Unidentified callers are not allowed are not allowed to leave message for the user. the time that is taken by samples B. the time that is taken to encoded/compression speech D.QUESTION NO: 245 Which statement about the Cisco Unity Connection message quota enforcement policies when a mailbox has exceeded the send/receive quota is incorrect? A. the user is unable to send messages. the time that is required interface E. Which three object classes are extended by the Cisco Unity schema extension process? (Choose three) A. D. Message from other users generate nonedelivery receipts to the senders.

To change the maximum recording length.000 (1 minute) to 36. group F. Traffic policing propagates burst.com/en/US/docs/voice_ip_comm/connection/7x/administra tion/guide/7xcucsag22 0.000 milliseconds (5 minutes) in length. contact Answer: A. 15 minutes D.html#wp1050330 QUESTION NO: 249 which two characteristics about traffic policing on Cisco IOS VoIP gateways are correct? (Choose Two) A. http://www.D. B. computer C. domain D.000. 30 minutes Answer: A Explanation: Maximum Recording Length—Indicates the maximum length allowed for system broadcast messages. enter a number from 60.B. organizational unit E. senders can record messages up to 300. 5 minutes B.000 (60 minutes) milliseconds.F Explanation: QUESTION NO: 248 What is the default maximum recording length that is allowed for system broadcast messages on a Cisco Unity Connection server? A. . By default. 20 minutes E. Traffic policing buffers and re-marks excess packets above the committed rates.cisco. 10 minutes C.

729A are both high-complexity codecs.729A is a medium-complexity codec. G. G. Station Start Media transmission message D. Traffic policing D. The difference between the G.C Explanation: QUESTION NO: 250 token values are configured in bits per second. E. Station IP Port Message B.729 is a low-complexity codec.C. while G.729 and G.while G.729 Annex-B codec is that the G.729A Annex-B is a medium complexity variant of G.729 Annex-B with slightly lower voice quality. while G. G.729 and G. Traffic policing F. could introduce delays because of deep queues.729 Annex-B is a high complexity algorithm. Traffic policing traffic. QUESTION NO: 251 Which SCCP message is used to instruct to an SCCP IP Phone the remote IP address and port number to send RTP packets? A. Answer: C Explanation: . Which statement about the G. D.729B both provice built-in VAD C.729 is a low-complexity codec. Station Call information message. G. B. Station Open Receive Channel message C.729A is mediumcomplexity codec. E. Traffic policing Answer: B. Station Open Logical Channel message. Answer: D Explanation: G.729 codec is correct? A.729 Annex-B codec provides built-in IETF voice activity detection (VAD) and Comfort Noise Generation (CNG).729 is a high-complexity codec . is applicable to both inbound and outbound is an inbound-only concept. E. G.729A is a high-complexity codec.729A and G. and G.

G. E. B. LPM will not do anything. administrator must manually remove access files in the common partition.729 is a low-complexity codec. Answer: D Explanation: QUESTION NO: 254 Two H. The error alert that is shown in the exhibit is seen in the “Event ViewerApplication Log” on Cisco Unified Presence.729A is a mediumcomplexity codec. LPM will not do any thing administrator must manually remove access files in the active partition. while G.323 gateways are engaged in an active call. G. Answer: D Explanation: QUESTION NO: 253 Refer to the Exhibit.729A is a mediumcomplexity codec. G. D. G. while G. 2 B. Which action will be performed by Cisco LPM tool in response to the alert? A. 3 . G. C. while G. How many RTP and RTCP packet streams exist between these two gateways? A.729B both provide built-in VAD. C. D. LPM will purge some of the trace and core files until 50 percent of the disk space is available. LPM will Purge trace and core files until disk usage is below the configured low watermark.729A is a high-complexity codec. E.729A and G.729A are both high-complexity codecs.729 is a low-complexity codec.729 is a high-complexity codec.729 codec is correct? A.729 and G. LPM will purge all trace files and core files.QUESTION NO: 252 which statement about the G. B.

C. 6 Answer: A . 5 E. 4 D.