CCIE Voice 350-030 - GRATIS EXAM...Refer to the exhibit. 4 Traffic flows from the IP phone that is...

CCIE Voice 350-030

Number: 1Passing Score: 800Time Limit: 120 minFile Version: 1.0

http://www.gratisexam.com/

Cisco 350-030 - updated with Explanations and removal of Actual Test spam

CCIETM Voice Written

Version: 20.0Cisco 350-030 Exam

Topic 1, Volume A

Exam A

QUESTION 1What are two advantages of multicast technologies? (Choose two.)

A. Denial of service attacks in the network are prevented.B. They eliminate multipoint applications.C. They reduce traffic by delivering a separate stream of information to each corporate recipient or home

environment, which reduces bandwidth.D. They control network traffic and reduce server and CPU load.E. They eliminate traffic redundancy.

Correct Answer: DESection: (none)Explanation

Explanation/Reference:Explanation:

QUESTION 2Which two descriptions apply to the Calling Search Space function in Cisco Unified Communications Manager?(Choose two.)

A. It defines which numbers are available for a device to call.B. It provides a group of dial patterns to look through when making a call.C. Within a partition, each CSS has a directory number.D. It defines route patterns and directory numbers from which calls can be received.E. It defines the search for directory numbers in assigned partitions according to dial patterns.

Correct Answer: AESection: (none)Explanation


QUESTION 3Which two statements apply to the partitions function in Cisco Unified Communications Manager? (Choosetwo.)

A. When a directory number or route pattern is placed into a certain partition, this creates a rule for who cancall that device or route list.

B. A partition is a logical grouping of directory numbers and route patterns that have similar reachabilitycharacteristics.

C. Calling Search Spaces are assigned to partitions.D. A directory number may appear in only one partition.E. Within the partition, each CSS has a directory number.

Correct Answer: ABSection: (none)Explanation


QUESTION 4Which three statements are true about multicast IGMP snooping? (Choose three.)

A. When a host in a multicast group sends an IGMP leave message, only that port is deleted from themulticast group.

B. An IP multicast stream to the IP host can be stopped only by an IGMP leave message.C. IGMP snooping does not examine or snoop Layer 3 information in packets that are sent between the hosts

and the router.D. When the switch hears the IGMP host report from a host for a particular multicast group, the switch adds

the host's port number to the associated multicast table entry.E. IGMP control messages are transmitted as IGMP multicast packets so that they can be distinguished from

normal multicast data at Layer 2.F. A switch that is running IGMP snooping examines every multicast data packet to verify whether it contains

any pertinent IGMP "must control" information.

Correct Answer: ADFSection: (none)Explanation


QUESTION 5Which three options are valid SCCP call states sent to an IP phone?

A. Ring OffB. On HookC. Call TransmitD. ConnectedE. DisconnectedF. In Use Remotely

Correct Answer: BDFSection: (none)Explanation

Explanation/Reference:Explanation:1—Off Hook2—On Hook3—Ring Out4—Ring In5—Connected6—Busy7—Line In Use8—Hold9—Call Waiting10—Call Transfer11—Call Park12—Call Proceed13—In Use Remotely14—Invalid Number

fromhttp://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080624977.shtml

QUESTION 6Which three statements are true about Cisco Discovery Protocol? (Choose three.)

A. It is an excellent tool for displaying the interface status on switches.B. It works on top of the network layer and data link level.C. It uses a multicast packet with a destination MAC address of 01-00-CC-CC-CC.D. The platform TLV (TLV type 0x0006) contains an ASCII character string that describes the hardware

platform of the device.E. You can use the CDP timer feature to change update times. The default is 60 seconds.F. It uses a broadcast packet with a destination MAC address of 01-00-CC-CC-CC.

Correct Answer: ADESection: (none)Explanation



QUESTION 7Which two of the following are functions of DHCP snooping? (Choose two.)

A. relies on already discovered trusted and untrusted portsB. dynamic ARP inspectionC. defines trusted and untrusted portsD. uses existing binding tablesE. builds a binding tableF. automatically builds ACLs

Correct Answer: CESection: (none)Explanation

Explanation/Reference:Explanation:The DHCP snooping feature determines whether traffic sources are trusted or untrusted.The DHCP snooping binding database is also referred to as the DHCP snooping bindingtable.

fromhttp://www.cisco.com/en/US/docs/switches/lan/catalyst6500/ios/12.2SXF/native/configuration/guide/snoodhcp.html#wp1120427

QUESTION 8

Refer to the exhibit.

4

Traffic flows from the IP phone that is connected to SW1 to the IP phone on SW2. If the trust boundary hasbeen extended to the IP phone on SW1, in what two places will traffic be marked and classified so that theproper QoS settings may be carried through the network? (Choose two.)

A. IP phone attached to SW1B. SW1 ingress portC. R1 ingress portD. SW1 egress portE. R1 egress port

Correct Answer: BCSection: (none)Explanation


Not ip Phone? Priority extend trust - trust device cisco-phone

QUESTION 9Refer to the exhibit.

Which gatekeeper mechanism prevents the gatekeeper from using all the resources on either gateway 1 orgateway 2 when sending calls to zones SE and NW?

A. bandwidth remoteB. resource availability indicatorC. bandwidth totalD. bandwidth zoneE. lrq immediate advanceF. ras timeout brq

Correct Answer: BSection: (none)Explanation


QUESTION 10When implementing a Cisco Unified Communications Manager solution over an MPLS WAN, which two rulesmust be observed to prevent overrunning the priority queue? (Choose two.)

A. RSVP will transparently pass application IDs from the customer network across the MPLS WAN.B. The media streams must be the same size in both directions.C. Only the connection to the MPLS WAN where the Cisco Unified Communications Manager resides must be

enabled as a CE device.D. The media has to be symmetrically routed.E. If the CE is under corporate control, it may support either topology-aware or measurement- based CAC.

Correct Answer: BDSection: (none)Explanation


QUESTION 11How is fax pass-through traffic treated over IP WAN connections that use the G.729 codec?

A. The fax traffic is demodulated and sent with VAD and echo chancellor disabled.B. When the TGW detects the CED tone from the fax machine that has been contacted, the TGW changes to

the G.711 codec with echo chancellor and VAD disabled.C. When the OGW detects the CED tone from the fax machine that is making the call, the OGW is informed

by the contacted device of the Cisco NSF features and switches to the G.711 codec with VAD disabled.D. The contacting fax machine sends a TCF message to the contacted fax machine and waits for a CFR

message. When the CFR message is received, the fax tones sent by the contacting fax machine cause theOGW to send an NSF message to the TGW, instructing it to switch to the

E. 711 codec with echo chancellor and VAD disabled.


Explanation/Reference:Explanation:Once a terminating gateway (TGW) detects a CED tone from a called fax machine, theTGW exchanges the voice codec that was negotiated during the voice call setup for a G.711codec and turns off EC and VAD.

fromhttp://www.cisco.com/en/US/docs/ios/12_3/vvf_c/cisco_ios_fax_services_over_ip_application_guide/fxappdoc.html


How many simultaneous G.729 calls can be established between sites SJ and RTP?

A. 4B. 5C. 6D. 8E. 12

Correct Answer: CSection: (none)Explanation

Explanation/Reference:Explanation:G729 = 8kbit. 8 x 2 = 16. Interzone bandwith = 96. 96/16 = 6

QUESTION 13

Company Alpha has a central office and a branch office that utilize a central call processing toplogy. Callsbetween the two sites are using the G.729 codec; calls within each site are using the G.711 codec.To conference an existing call between two phones at the central site with a phone at the remote office, whichtwo of the following are possible solutions? (Choose two.)

A. a software conference bridge that is configured in Cisco Unified Communications ManagerB. a software conference bridge that is configured in Cisco Unified Communications Manager and a HW

transcoderC. a hardware conference bridgeD. a hardware transcoder and a hardware conference bridgeE. No extra configurations required--phones automatically negotiate using the lowest common denominator

codec (G.729)


Explanation/Reference:Explanation:Software Ressources are not capable of transcoding, always HW required for this


You have been asked to edit the sample auto attendant script so that callers are prompted to press 1 for sales,

2 for service, or 3 for the directory. If callers select 3, they should hear the existing menu choices to dial byextension, dial by name, or transfer to the operator. What steps can you take to create this nested menu?

A. Drag a new Menu step from the palette and drop it on the Start step. Drag the existing Menu step and dropit on Output 3 of the new Menu.

B. Drag a new Menu step from the palette and drop it on the existing Menu step. This will make the existingMenu subordinate to the new Menu.

C. Drag a new Menu step from the palette and drop it on the existing Menu step. Drag the existing Menu stepand drop it on Output 3 of the new Menu.

D. Delete the existing Menu. Drag a new Menu step from the palette and drop it on the Set prefixPrompt=P[]step. Recreate the existing directory menu as the third option of the new Menu step.



QUESTION 15Which two of these are possible reasons why a JTAPI subsystem might have the status PARTIAL_SERVICE?(Choose two.)

A. Cisco Unified Contact Center is not able to resolve the host name of Cisco Unified CommunicationsManager.

B. A referenced CTI Route Point is not associated with the JTAPI user.C. The JTAPI user password is not correct.D. There is an error in one of the scripts being loaded.E. The CTI Manager service is not running on Cisco Unified Communications Manager.



Refer to the Cisco Unified CCX trace files to determine what did not initialize. Verify that all CTI ports and CTI route points are associated with the JTAPI user in Cisco Unified CM. Verify that the Cisco Unified CM and JTAPI configuration IP addresses match. Verify that the Cisco Unified CM JTAPI user has control of all the CTI ports and CTI route points. Verify that the application file was uploaded to the repository using the Repository Manager.

fromhttp://docwiki.cisco.com/wiki/JTAPI_subsystem_is_in_partial_service

QUESTION 16Which three of these are mandatory sub-commands of the call-manager-fallback command and will help an IPphone register to an IOS router in SRST mode? (Choose three.)

A. access-codeB. dialplan-patternC. ip source-addressD. keepaliveE. max-dnF. max-ephones

Correct Answer: CEFSection: (none)Explanation



You are debugging a problem on a SIP network and have run the debug ccsip messages command. One of themessages returned is shown in the exhibit. What information will the server return to the caller?

A. the acceptable media typeB. a list of acceptable media typesC. a list of acceptable formatsD. a correct directory numberE. an acceptable language code



QUESTION 18Which type of SIP responses would indicate that a server encountered an error in attempting to complete a SIPrequest?

A. 1xxB. 3xxC. 4xxD. 5xxE. 6xx

Correct Answer: DSection: (none)Explanation


5xx—Server Failure Responses500 Server Internal Error 501 Not Implemented: The SIP request method is not implemented here 502 Bad Gateway 503 Service Unavailable 504 Server Time-out 505 Version Not Supported: The server does not support this version of the SIP protocol 513 Message Too Large 580 Precondition Failure

http://en.wikipedia.org/wiki/List_of_SIP_response_codes

QUESTION 19 11

To hide its identity when initiating calls, SIP Phone B requests that Server B place its calls for it.

What kind of device is Server B?

A. proxyB. redirectC. registrarD. user agent clientE. user agent server

Correct Answer: ASection: (none)Explanation


Proxy Server: The proxy server is an intermediary entity that acts as both a server and a client for the purposeof making requests on behalf of other clients. A proxy server primarily plays the role of routing, meaning that itsjob is to ensure that a request is sent to another entity closer to the targeted user. Proxies are also useful forenforcing policy (for example, making sure a user is allowed to make a call). A proxy interprets, and, ifnecessary, rewrites specific parts of a request message before forwarding it.

fromhttp://www.cisco.com/web/about/ac123/ac147/archived_issues/ipj_6-1/sip.html

QUESTION 20Which of the following three messages could be sent by the UAC in response to the 180 Ringing? (Choosethree.)

A. PRACKB. ACKC. BYED. CANCELE. INVITE

Correct Answer: ABDSection: (none)Explanation


fromhttp://www.cisco.com/en/US/tech/tk652/tk701/technologies_configuration_guide_chapter09186a00800eadfa.html(Prack is not listed here but the others make no sense and prack is known according to google etc)

QUESTION 21Which three attributes correctly describe aspects of MGCP? (Choose three.)

A. peer-to-peerB. Master/SlaveC. call preservation on gateway failover from one Cisco Unified Communications Manager server to anotherD. communication with Cisco Unified Communications Manager handled via a proxy serverE. centralized dial plan managementF. intelligent endpoints

Correct Answer: BCE

Section: (none)Explanation


MGCP (defined under RFC 2705) is a master/slave protocol that allows a call control device (such as CiscoCallManager) to take control of a specific port on a gateway. This has the advantage of centralized gatewayadministration and provides for largely scalable IP Telephony solutions. With this protocol, the CiscoCallManager knows and controls the state of each individual port on the gateway. It allows complete control ofthe dial plan from Cisco CallManager, and gives CallManager per-port control of connections to the publicswitched telephone network (PSTN), legacy PBX, voice mail systems, plain old telephone service (POTS)phones, and so forth. This is implemented with the use of a series of plain-text commands sent over UserDatagram Protocol (UDP) port 2427 between the Cisco CallManager and the gateway. A list of the possiblecommands and their functions is provided later in this document.

Call preservation—calls are maintained during failover and failback

fromhttp://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a00806fedbe.shtml

QUESTION 22In a VoIP deployment, which two protocols satisfy the following three requirements? (Choose two.)

Requirement 1: the protocol has a mechanism for a centralized dial-plan

Requirement 2: the endpoints are considered to be unintelligent

Requirement 3: the protocol is text-based

A. SIPB. H.323C. MGCPD. SCCP

Correct Answer: CDSection: (none)Explanation


QUESTION 23When implementing PRI backhaul for an MGCP gateway and Cisco Unified Communications Manager, theQ.921 data-link protocol is terminated on which device?

A. Cisco Unified Communications ManagerB. MGCP gatewayC. signaling link terminalD. the IP end device, such as an IP phone



MGCP PRI backhaul terminates all ISDN PRI Layer 2 (Q.921) signaling functions on the MGCP gateway while,at the same time, packaging all the ISDN PRI Layer 3 (Q.931) signaling information into packets fortransmission to Cisco Unified Communications Manager through an IP tunnel over a TCP connection. Thisensures the integrity of the Q.931 signaling information that passes through the network for managing IPtelephony devices. A rich set of user-side and network-side ISDN PRI calling functions is supported by MGCPPRI backhaul.

http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/interop/intcnf3.html

QUESTION 24What 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.



QUESTION 25On which gateway or gatekeeper is the IOS command call-rsvp-sync resv-timer 10 used to set the timer?

A. originating VoIP gateway for completing RSVP reservation setups in 10 secondsB. originating and terminating VoIP gateway for completing RSVP reservation setups in 10 secondsC. terminating VoIP gateway for completing RSVP reservation setups in 10 secondsD. VoIP gatekeeper for completing RSVP reservation setups in 10 seconds



A timer can be set by using the call rsvp-sync serv-timer command to limit the number of seconds that theterminating gateway waits for bandwidth reservation setup before proceeding with the call setup or releasingthe call, depending on the configured QoS level in the dial peers.

http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/cisco_ios_voice_configuration_library_glossary/vclqos.html

QUESTION 26If the bandwidth total default 64 command is configured in a gatekeeper, then what is true of that gatekeeper?

A. it will admit up to 64 calls, regardless of codec usedB. it will not admit any calls because all calls initially account of 128 kb/sC. it will admit a minimum of four calls using the G.729 codecD. it will admit up to four calls using the G.729 codec

E. it will admit a G.711 call in one direction only, since 64 is half of 128 kb/s



G729 = 8kbit/sec. x 2 (gatekeeper takes double bandwith to compensate for in/out bound). 64 / 16 = 4 totalcalls.

QUESTION 27If enabled, the RSVP for LLQ feature will assign which two types of flows to the priority queue? (Choose two.)

A. all RSVP bandwidth requestsB. voice flows generated from Cisco IOS applicationsC. voice flows generated from third-party applications, such as Microsoft NetMeetingD. all traffic marked DSCP EFE. all traffic marked CoS 5



RSVP is a network-control protocol that provides a means for reserving network resources—primarilybandwidth—to guarantee that applications sending end-to-end across networks achieve the desired QoS.

RSVP enables real-time traffic (which includes voice flows) to reserve resources necessary for low latency andbandwidth guarantees.

Voice traffic has stringent delay and jitter requirements. It must have very low delay and minimal jitter per hop toavoid degradation of end-to-end QoS. This requirement calls for an efficient queueing implementation, such aslow latency queueing (LLQ), that can service voice traffic at almost strict priority in order to minimize delay andjitter.

RSVP uses WFQ to provide fairness among flows and to assign a low weight to a packet to attain priority.However, the preferential treatment provided by RSVP is insufficient to minimize the jitter because of the natureof the queueing algorithm itself. As a result, the low latency and jitter requirements of voice flows might not bemet in the prior implementation of RSVP and WFQ.

RSVP provides admission control. However, to provide the bandwidth and delay guarantees for voice traffic andget admission control, RSVP must work with LLQ. The RSVP Support for LLQ feature allows RSVP to classifyvoice flows and queue them into the priority queue within the LLQ system while simultaneously providingreservations for nonvoice flows by getting a reserved queue

fromhttp://www.cisco.com/en/US/docs/ios/12_2/qos/configuration/guide/qcfsig.html#wpxref47981

QUESTION 28Which of these features are supported in RSVP Support for LLQ? (Choose three.)

A. LLQ Support on TunnelsB. Guaranteed Quality of Service

C. Reserve resources for Low Latency and bandwidth guaranteesD. LLQ on Frame Relay and ATM PVCsE. Controlled-Load Network Element Service

Correct Answer: BCESection: (none)Explanation


RFC 221, Controlled-Load Network Element ServiceRFC 2212, Specification of Guaranteed Quality of ServiceBandwidth guarantee. The RSVP reservation, if accepted, will guarantee that those reserved resources willcontinue to be available while the reservation is in place.

http://www.cisco.com/en/US/docs/ios/12_2/qos/configuration/guide/qcfsig.html#wpxref47981

QUESTION 29Users are complaining that the music on hold marketing files for this month are not being played when usersare placed on hold. Which three of these do you need to verify? (Choose three.)

A. the IP voice media streaming application has been stopped and restartedB. a new directory has been created for the new media filesC. users have selected the correct MoH files for customer callsD. the new music files are in the correct format to be used with Cisco Unified Communications ManagerE. the location of the new music files is what the MoH server expects



Users cannot specify the MOH file and a new directory is not required. Only the above is correct.

QUESTION 30Which of these statements correctly describes the logic for selecting MoH servers and MoH audio streams?

A. The audio stream and audio server used will be selected according to the configuration of the phone beingplaced on hold.

B. The audio stream and audio server used will be selected according to the configuration of the phone whichis being used to place a caller on hold.

C. The audio stream will be selected according to the configuration of the phone which is being used to place acaller on hold, and the audio server used will be selected according to the configuration of the phone beingplaced on hold.

D. The audio stream will be selected according to the configuration of the phone being placed on hold and theaudio server used will be selected according to the configuration of the phone which is being used to placea caller on hold.


Explanation/Reference:

Explanation:

QUESTION 31Which two conditions will result in an H.323 gatekeeper receiving an ARQ from a registered H.323 endpoint?(Choose two.)

A. A remote zone endpoint initiates a call.B. A local zone endpoint requests permission to admit an incoming call.C. A remote zone endpoint sends keepalive to ensure registration continuity.D. A remote zone gatekeeper initiates a call.E. A local zone endpoint initiates a call.F. A local zone endpoint sends keepalive to ensure registration continuity.

Correct Answer: BESection: (none)Explanation


QUESTION 32Two H.323 gateways are engaged in an active call. How many RTP and RTCP packet streams exist betweenthese two gateways?

A. 2B. 3C. 4D. 5E. 6



QUESTION 33On a Cisco IOS MGCP gateway that is registered to Cisco Unified Communications Manager, which MGCPmessage is initiated by the gateway?

A. RQNTB. NTFYC. EPCFD. CRCXE. SETUP



EPCF EndpointConfiguration—specifies the encoding of the signals that will be received by the endpoint.

RQNT NotificationRequest—requests the gateway to send notifications upon the occurrence of specifiedevents in an endpoint.

NTFY Notify—sent by the gateway in compliance with RQNT when a triggering event occurs.

CRCX CreateConnection—creates a connection between two endpoints.

MDCX ModifyConnection—modifies the characteristics of a gateway's "view" of a connection. This "view" ofthe call includes both the local connection descriptor as well as the remote connection descriptor.

DLCX DeleteConnection (from the Call Agent)—terminates a connection. As a side effect, it collectsstatistics on the execution of the connection.DeleteConnection (from the gateway)—issued by the media gateway to clear a connection, forexample because it has lost the resource associated with the connection, or because it hasdetected that the endpoint no longer is capable or willing to send or receive voice.DeleteConnection (multiple connections, from the Call Agent)—used by the Call Agent to deletemultiple connections at the same time. The command can be used to delete all connections thatrelate to a Call for an endpoint or terminate in a given endpoint.

AUEP AuditEndpoint—used by the call agent to find out the status of a given endpoint.

AUCX AuditConnection—used by the Call Agent to retrieve the parameters attached to a connection.

RSIP RestartInProgress—used by the gateway to signal that an endpoint, or a group of endpoints, is putin-service or out-of-service.

fromhttp://www.cisco.com/en/US/products/hw/switches/ps1925/products_maintenance_guide_chapter09186a008008745c.html#xtocid10

QUESTION 34On a Cisco IOS MGCP gateway that is registered to Cisco Unified Communications Manager, which fourMGCP messages are initiated by Cisco Unified Communications Manager? (Choose four.)

A. AUEPB. MDCXC. RQNTD. NTFYE. RSIPF. CRCX

Correct Answer: ABCFSection: (none)Explanation












QUESTION 35On a Cisco IOS MGCP gateway that is registered to Cisco Unified Communications Manager, which MGCPmessage could be initiated by either Cisco Unified Communications Manager or the gateway?

A. CRCXB. RQNTC. DLCXD. AUEPE. RSIP








DLCX DeleteConnection (from the Call Agent)—terminates a connection. As a side effect, it collectsstatistics on the execution of the connection.DeleteConnection (from the gateway)—issued by the media gateway to clear a connection, forexample because it has lost the resource associated with the connection, or because it hasdetected that the endpoint no longer is capable or willing to send or receive voice.DeleteConnection (multiple connections, from the Call Agent)—used by the Call Agent to delete

multiple connections at the same time. The command can be used to delete all connections thatrelate to a Call for an endpoint or terminate in a given endpoint.






What is the maximum number of inbound calls to 2001 before a Cisco Unified Communications ManagerExpress system returns a user busy tone to any additional calls?

A. 3B. 4C. 5D. 6E. 7




What is the maximum number of inbound calls to ephone 1 before a Cisco Unified Communications ManagerExpress system returns a user busy tone to any additional calls?

A. 3B. 4C. 5D. 6E. 7




What is the maximum number of calls that are supported on ephone 2?

A. 3B. 4

C. 5D. 6E. 8




What is the maximum number of inbound calls to 2001 before a Cisco Unified Communications ManagerExpress system returns a user busy tone to any additional calls?

A. 4B. 5C. 6D. 7E. 8



QUESTION 40Which two analog voice interfaces support ground-start? (Choose two.)

A. FXSB. E&M Type IC. E&M Type IID. E&M Type IVE. FXO



Also, Cisco only supports E&M Type 1-3 and 5. NOT 4.

fromhttp://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00803736c1.shtml

QUESTION 41Which signaling method cannot solve the FXO disconnect problem?

A. power denialB. tone-based supervisory disconnectC. pulse dialD. ground-start signalingE. battery reversal



fromhttp://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800ae2d1.shtml

QUESTION 42Which three signaling types are not used by analog E&M circuits as start dial supervision protocols? (Choosethree.)

A. delay dialB. wink-startC. ground-startD. wink-start Feature Group DE. immediate-startF. pulse dial

Correct Answer: CDFSection: (none)Explanation


On E&M circuits, the three main Start Dial Supervision protocols are:Immediate StartWink StartDelay Dial

fromhttp://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a0080094ab9.shtml

QUESTION 43Which three are valid T1 CAS types? (Choose three.)

A. E&M signalingB. semicompelled signalingC. loop-start signalingD. line signalingE. Group 1 signalingF. ground-start signaling

Correct Answer: ACFSection: (none)Explanation


E&M Signaling is typically used for trunk lines. The signaling paths are known as the E-lead and the M-lead.Descriptions such as Ear and Mouth were adopted to help field personnel determine the direction of a signal ina wire. E&M connections from routers to telephone switches or to PBXs are preferable to FXS/FXOconnections because E&M provides better answer and disconnect supervision.

Loopstart signaling is one of the simplest forms of CAS signaling. When a handset is picked up (the telephonegoes off-hook), this action closes the circuit that draws current from the telephone company CO and indicates achange in status, which signals the CO to provide dial tone. An incoming call is signaled from the CO to thehandset by sending a signal in a standard on/off pattern, which causes the telephone to ring.

Groundstart signaling is very similar to loopstart signaling in many regards. It works by using ground andcurrent detectors that allow the network to indicate off-hook or seizure of an incoming call independent of theringing signal and allow for positive recognition of connects and disconnects. For this reason, ground startsignaling is typically used on trunk lines between PBXs and in businesses where call volume on loop start linescan result in glare.

fromhttp://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800e2560.shtml

QUESTION 44Which R2 signaling element passes address information such as calling- and called-party numbers?

A. pulse signalingB. delay dial signalingC. line signalingD. interregister signalingE. out-of-band signaling



Interregister Signaling (Call Setup Control Signals )The concept of address signaling in R2 is slightly different than that used in other CAS systems. In R2signaling, the exchanges are considered registers and the signaling between these exchanges is calledinterregister signaling. Interregister signaling uses forward and backward in-band multifrequency signals ineach time slot to transfer called and calling party numbers, as well as the calling party category.

fromhttp://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800943c2.shtml

QUESTION 45How many frames are contained in one multiframe within an SF format?

A. 4B. 8C. 15D. 16E. 30F. 32



QUESTION 46How many channels on a voice E1 circuit are used to carry PCM-encoded voice traffic?

A. 16B. 28C. 29D. 30E. 31



QUESTION 47According to the IEEE 802.3af PoE standard, what is the maximum power (in watts) that is delivered to apower-consuming device?

A. 6.3B. 14.5C. 15.4D. 20E. 22.5F. 25.4



The original IEEE 802.3af-2003[2] PoE standard provides up to 15.4 W of DC power (minimum 44 V DC and350 mA) to each device. Only 12.95 W is assured to be available at the powered device as some power isdissipated in the cable.

fromhttp://en.wikipedia.org/wiki/Power_over_Ethernet

QUESTION 48What is the complete name of LLDP-MED, an enhancement to the vendor-neutral LLDP that is supported onCisco switches?

A. Link Layer Discovery Protocol-Media Endpoint DiscoveryB. Link Layer Discovery Protocol-Media Enhancement DeliveryC. Link Layer Discovery Protocol-Media Enhancement DiscoveryD. Link Layer Discovery Protocol-Multiple Enhancement DeliveryE. Link Layer Discovery Protocol-Multiple Endpoint Discovery



Media Endpoint Discovery is an enhancement of LLDP, known as LLDP-MED, that provides the followingfacilities:

fromhttp://en.wikipedia.org/wiki/Link_Layer_Discovery_Protocol

QUESTION 49Which of these is not a valid switchback method for SCCP hardware conference bridges?

A. immediateB. neverC. gracefulD. guardE. uptime



switchback method {graceful | guard [timeout-value] | immediate | uptime uptime-value}

fromhttp://www.cisco.com/en/US/docs/ios/12_3/vvf_c/interop/intcnf2.html

QUESTION 50Which two are valid switchover methods for SCCP hardware conference bridges? (Choose two.)

A. immediateB. guard

C. uptime delayD. schedule timeE. gracefulF. never



switchback method {graceful | guard [timeout-value] | immediate | uptime uptime-value}


QUESTION 51What is the default switchback method for an SCCP hardware transcoder when a higher-priority Cisco UnifiedCommunications Manager becomes available again?

A. gracefulB. immediateC. uptimeD. neverE. guard

Correct Answer: ESection: (none)Explanation


Router(config-sccp-ccm)# switchback method graceful (Optional) Sets the switchback method to use when the primary or higher priority Cisco UnifiedCommunications Manager becomes available again. •Default is guard, with a timeout value of 7200 seconds.


QUESTION 52Which telephony signaling type cannot be configured for a Cisco IOS MGCP gateway on Cisco UnifiedCommunications Manager?

A. T1 PRIB. analog FXOC. analog E&MD. T1 CAS E&M delay dialE. ISDN BRI



The following types of interfaces on the gateway are supported:

•FXS analog interfaces—For connecting to the PSTN or analog phones •FXO analog interfaces—For connecting to the PSTN or PBXs •T1 CAS digital interfaces—For connecting to the PSTN or PBXs •T1 and E1 PRI digital interfaces—For connecting to PBXs and central offices (COs)


QUESTION 53Which two codecs provide built-in VAD? (Choose two.)

A. G.711 mu-lawB. G.722-64KC. G.723.1 Annex AD. G.726E. G.729F. G.729 Annex B

Correct Answer: CFSection: (none)Explanation

Explanation/Reference:Explanation:There are two versions of G.723.1 called Annex-A and non Annex-A. These versions do not interoperate.G.723.1 Annex-A includes a built-in IETF VAD algorithm and CNG.G.729 Annex-B codec provides built-in IETF voice activity detection (VAD) and Comfort Noise Generation(CNG).

QUESTION 54Which statement about the G.729 codec is correct?

A. G.729 and G.729A are both high-complexity codecs.B. G.729A and G.729B both provide built-in VAD.C. G.729 is a low-complexity codec, while G.729A is a high-complexity codec.D. G.729 is a high-complexity codec, while G.729A is a medium-complexity codec.E. G.729 is a low-complexity codec, while G.729A is a medium-complexity codec.



Medium Complexity (4 calls /dsp)

High Complexity ( 2 calls /dsp)

G.711 (a-law and m -law) G.728

G.726 (all versions) G.723 (all versions)

G.729a, G.729ab (G.729aAnnexB)

G.729, G.729b (G.729-AnnexB)

Fax-relay Fax-relay

fromhttp://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a00800b6710.shtml

QUESTION 55Which three codecs are considered to have medium complexity on Cisco IOS voice gateways? (Choose three.)

A. G.711 a-lawB. G.711 mu-lawC. G.723D. G.728E. G.729F. G.729 Annex A with Annex B

Correct Answer: ABFSection: (none)Explanation







G.729, G.729b (G.729-AnnexB)

Fax-relay Fax-relay


QUESTION 56Which three codecs are considered to have high complexity on Cisco IOS voice gateways? (Choose three.)

A. G.711 a-lawB. G.711 mu-lawC. G.723D. G.728E. G.729F. G.729 Annex B

Correct Answer: CDFSection: (none)

Explanation







G.729, G.729b (G.729-AnnexB)

Fax-relay Fax-relay


QUESTION 57Which statement about the G.729 codec is correct?

A. G.729 Annex A is a high-complexity codec.B. G.729 Annex A and G.729 do not interoperate with each other.C. G.729 Annex A with Annex B is a pre-IETF-standard format.D. G.729 Annex A with Annex B and G.729 Annex B can interoperate with each other only through a

transcoder.E. The Cisco IOS configuration option of "g729r8" uses G.729 Annex A when medium complexity is defined on

the voice card.








G.729, G.729b (G.729-AnnexB)

Fax-relay Fax-relay


QUESTION 58On a Cisco IOS MGCP PRI gateway, what is the maximum configurable length of time for a scheduledswitchback to a higher-priority Cisco Unified Communications Manager?

A. 6 hoursB. 12 hoursC. 18 hoursD. 24 hoursE. 48 hours



#ccm-manager switchback uptime-delay ? <1-1440> Delay time (minutes)

1440/60 = 24

QUESTION 59Which of these is an invalid switchback method for a Cisco IOS MGCP PRI gateway in case a higher-priorityCisco Unified Communications Manager returns to active service?

A. guardB. gracefulC. immediateD. schedule-timeE. uptime delay



ccm-manager switchback {graceful | immediate | schedule-time hh:mm | uptime-delay minutes}


QUESTION 60What is the default switchback method for a Cisco IOS MGCP PRI gateway when a higher-priority CiscoUnified Communications Manager becomes available again?

A. gracefulB. immediateC. uptime delayD. neverE. schedule time



Configures switchback mode for returning control to the primary Cisco Unified Communications Manager.

•Default is graceful.

QUESTION 61Which statement about the Media Resource Group on Cisco Unified Communications Manager is correct?

A. Different types of media resources cannot be grouped into the same Media Resource Group.B. A Media Resource Group contains a prioritized list of media resources.C. The default Media Resource Group is defined in the service parameters of Cisco Unified Communications

Manager.D. Once a media resource is associated with a Media Resource Group, it is no longer eligible to be associated

with another Media Resource Group.E. The Media Resource Group configuration page allows administrators to choose whether to use multicast for

MOH audio.



QUESTION 62Which statement about MRGL on Cisco Unified Communications Manager is incorrect?

A. MRGL can be assigned to devices at the device level, device pool level, or both.B. MRGL contains a prioritized list of Media Resource Groups.C. Media resources that are not contained in any Media Resource Groups are not used by MRGL.D. MRGL can contain a single Media Resource Group.E. When a call is placed on hold, the MRGL of the device that put the call on hold determines which MOH

server is used to play music to the held device.



The last MRGL is the default MRGL. A media resource that is not assigned to an MRG is automaticallyassigned to the default MRGL. The default MRGL is always searched and it is the last resort if no resources areavailable in the device-based MRGL and the device pool MRGL or if no MRGLs are configured at any level.

fromhttp://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a008020f198.shtml

QUESTION 63Cisco Unified Communications Manager Server A and Cisco Unified Communications Manager Server B are inthe same cluster. The cluster has a total of four registered conference bridges: two software conferencebridges (one from each server) and two Cisco IOS hardware conference bridges. All four conference bridgesare registered to Server B as the primary call-processing node and Server A as the backup. If an administratoraccidentally deactivated the Cisco IP Voice Media Streaming Application service on Server B, what will happento the conference resources in the cluster?

A. All four conference bridges will register to Server A.B. The Server B software bridge will deregister; the other three bridges will register to Server A.C. The Server B software bridge will deregister; the other three bridges will remain registered to Server B.D. Both software bridges will deregister, and both hardware bridges will remain registered to Server B.E. Both software bridges will deregister, and both hardware bridges will register to Server A.



QUESTION 64Cisco Unified Communications Manager Server A and Cisco Unified Communications Manager Server B are inthe same cluster. The cluster has a total of four registered conference bridges: two software conferencebridges (one from each server) and two Cisco IOS hardware conference bridges. All four conference bridgesare registered to Server B as the primary call-processing node and Server A as the backup. If an administratoraccidentally deactivated the Cisco CallManager service on Server B, what will happen to the conferenceresources in the cluster?

A. All four conference bridges will register to Server A.B. The Server B software bridge will deregister; the other three bridges will register to Server A.C. The Server B software bridge will deregister; the other three bridges will remain registered to Server B.D. Both software bridges will deregister, and both hardware bridges will remain registered to Server B.E. Both software bridges will deregister, and both hardware bridges will register to Server A.




QUESTION 65Which string is not a valid route pattern on Cisco Unified Communications Manager?

A. 123@B. 123.

C. 123*D. 123$E. 123?



Character Description Examples

@ The at symbol (@) wildcard matches all NANPnumbers. Each route pattern can have only one @ wildcard.

The route pattern 9.@ routes or blocks all numbersrecognized by the NANP.

The following route patterns examples show NANPnumbers encompassed by the @ wildcard:

X The X wildcard matches any single digit in the range0 through 9.

The route pattern 9XXX routes or blocks allnumbers in the range 9000 through 9999.

! The exclamation point (!) wildcard matches one ormore digits in the range 0 through 9.

The route pattern 91! routes or blocks all numbersin the range 910 through91999999999999999999999.

? The question mark (?) wildcard matches zero ormore occurrences of the preceding digit or wildcardvalue.

The route pattern 91X? routes or blocks allnumbers in the range 91 through91999999999999999999999.

+ The plus sign (+) wildcard matches one or moreoccurrences of the preceding digit or wildcard value.

The route pattern 91X+ routes or blocks allnumbers in the range 9100 through91999999999999999999999.

[ ] The square bracket ([ ]) characters enclose a rangeof values.

The route pattern 813510[012345] routes or blocksall numbers in the range 8135100 through8135105.

- The hyphen (-) character, used with the squarebrackets, denotes a range of values.

The route pattern 813510[0-5] routes or blocks allnumbers in the range 8135100 through 8135105.

^ The circumflex (^) character, used with the squarebrackets, negates a range of values. It must be thefirst first character following the opening bracket ([). Each route pattern can have only one ^ character.

The route pattern 813510[^0-5] routes or blocks allnumbers in the range 8135106 through 8135109.

. The dot (.) character is used as a delimiter toseparate the Cisco CallManager access code fromthe directory number. Use this special character, with the discard digitsinstructions, to strip off the Cisco CallManageraccess code before sending the number to anadjacent system. Each route pattern can have only one . character.

The route pattern 9.@ identifies the initial 9 as theCisco CallManager access code in an NANP call.

* The asterisk (*) character can provide an extra digitfor special dialed numbers.

You can configure the route pattern *411 to provideaccess to the internal operator for directoryassistance.

# The octothorpe (#) character generally identifies theend of the dialing sequence. The # character mustbe the last character in the pattern.

The route pattern 901181910555# routes or blocksan international number dialed from within theNANP. The # character after the last 5 identifies

this as the last digit in the sequence.

fromhttp://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/3_1_2/ccmcfg/b03spchr.html

QUESTION 66Which Cisco Unified Communications Manager route pattern character represents zero or more occurrences ofthe previous digit or wildcard?

A. !B. +C. *D. .E. ?





















. The dot (.) character is used as a delimiter toseparate the Cisco CallManager access code from


the directory number. Use this special character, with the discard digitsinstructions, to strip off the Cisco CallManageraccess code before sending the number to anadjacent system. Each route pattern can have only one . character.




The route pattern 901181910555# routes or blocksan international number dialed from within theNANP. The # character after the last 5 identifiesthis as the last digit in the sequence.


QUESTION 67Which Cisco Unified Communications Manager route pattern character represents one or more occurrences ofdigits in the range of zero to nine?

A. !B. +C. *D. .E. ?




























QUESTION 68Which Cisco Unified Communications Manager route pattern character represents one or more occurrences ofthe previous digit or wildcard?

A. !B. +C. *D. .E. ?







The X wildcard matches any single digit in the range The route pattern 9XXX routes or blocks all

X 0 through 9. numbers in the range 9000 through 9999.





















Incoming calls that use this FXO port are hearing two rings before the destination endpoint, 1001, begins toring. What is a possible cause for this behavior?

A. connection plar opx in the current configuration is wrong; it should be replaced with connection plar.B. 1001 is configured for delayed ring.C. Caller ID is not provisioned by the telephone company on this FXO line.

D. FXO ports always ring twice before ringing the destination endpoint.



the two ring delay is for the caller ID to be received. This is a very common standards industry practice. Withouttwo rings first, the customer may not see the caller id right away.

from https://supportforums.cisco.com/thread/2021370

QUESTION 70What is the default method of handling an H.323 connection from unknown devices on Cisco UnifiedCommunications Manager?

A. Cisco Unified Communications Manager accepts incoming H.323 connections from unknown devices.B. Cisco Unified Communications Manager ignores incoming H.323 connections from unknown devices.C. Cisco Unified Communications Manager rejects incoming H.323 connections from unknown devices by

sending an H.225 reject.D. Cisco Unified Communications Manager rejects incoming H.323 connections from unknown devices by

sending an H.225 disconnect.E. Cisco Unified Communications Manager rejects incoming H.323 connections from unknown devices by

closing the TCP socket.



QUESTION 71What is the default method of handling an H.323 connection from unknown devices on a Cisco IOS H.323gateway?

A. A Cisco IOS H.323 gateway accepts incoming H.323 connections from unknown devices.B. A Cisco IOS H.323 gateway ignores incoming H.323 connections from unknown devices.C. A Cisco IOS H.323 gateway rejects incoming H.323 connections from unknown devices by sending an

H.225 reject.D. A Cisco IOS H.323 gateway rejects incoming H.323 connections from unknown devices by sending an

H.225 disconnect.E. A Cisco IOS H.323 gateway rejects incoming H.323 connections from unknown devices by closing the TCP

socket.



QUESTION 72

An H.225 call setup arrives at Cisco Unified Communications Manager for a directory number on an IP phonethat is engaged in an active conversation. If call waiting is disabled for this directory number and none of theCall Forward settings are defined, which H.225 disconnect reason code will be sent to the originating H.323gateway?

A. No Route To DestinationB. Normal Call ClearingC. Subscriber AbsentD. User BusyE. Network Busy



QUESTION 73When an H.225 call setup arrives at Cisco Unified Communications Manager for an IP phone directory numberwith a partition that is not reachable by the H.323 gateway calling search space, which H.225 disconnect reasoncode will be sent to the originating H.323 gateway?

A. No Route To DestinationB. Unallocated (Unassigned) NumberC. Number UnreachableD. Number Available But Out of ReachE. Network Busy



QUESTION 74When a call arrives from the PSTN on a Cisco IOS MGCP PRI gateway that is registered to Cisco UnifiedCommunications Manager, destined to an IP phone directory number with a partition that is not reachable bythe MGCP gateway calling search space, which event will take place?

A. Cisco Unified Communications Manager will send the call to the Call Forward Busy destination that isconfigured on the IP phone.

B. Cisco Unified Communications Manager will disconnect the call with an MGCP DLCX message.C. Cisco Unified Communications Manager will disconnect the call with a Q.931 cause of "No Route To

Destination".D. Cisco Unified Communications Manager will disconnect the call with a Q.931 cause of "Unallocated

(Unassigned) Number".E. Cisco Unified Communications Manager will disconnect the call with a cause code of 420, which means

"Bad Extension".



QUESTION 75A call arrives from the PSTN on a Cisco IOS MGCP PRI gateway that is registered to Cisco UnifiedCommunications Manager, destined for a directory number on an IP phone that is engaged in an activeconversation. If call waiting is disabled for this directory number and none of the Call Forward settings aredefined, which event will take place?

A. Cisco Unified Communications Manager will disconnect the call with an MGCP RSIP message.B. Cisco Unified Communications Manager will disconnect the call with an MGCP DLCX message with a

cause of "Busy".C. Cisco Unified Communications Manager will disconnect the call with a Q.931 cause of "User Busy".D. Cisco Unified Communications Manager will disconnect the call with a Q.931 cause of "Temporary Failure".E. Cisco Unified Communications Manager will disconnect the call with a cause code of 486, which means

"Busy Here".



QUESTION 76How many bits in an 802.1Q tagged Ethernet frame are used for 802.1p priority?

A. 3B. 4C. 5D. 6



Eight different classes of service are available as expressed through the 3-bit PCP field in an IEEE 802.1Qheader added to the frame. The way traffic is treated when assigned to any particular class is undefined and leftto the implementation. The IEEE however has made some broad recommendations:

PCP Network priority Acronym Traffic characteristics1 0 (lowest) BK Background0 1 BE Best Effort2 2 EE Excellent Effort3 3 CA Critical Applications4 4 VI Video, < 100 ms latency5 5 VO Voice, < 10 ms latency6 6 IC Internetwork Control7 7 (highest) NC Network ControlNote that the above recommendations were revised in IEEE 802.1Q-2005 and differ from the originalrecommendations found in IEEE 802.1D-2004.

from

http://en.wikipedia.org/wiki/IEEE_P802.1p


In this 802.1Q tagged Ethernet frame, which block of bits, labeled A, B, C, and D, is used as TPID?

A. AB. BC. CD. D



Tag Protocol Identifier (TPID) : a 16-bit field set to a value of 0x8100 in order to identify the frame as an IEEE802.1Q-tagged frame. This field is located at the same position as the EtherType/Length field in untaggedframes, and is thus used to distinguish the frame from untagged frames.

Priority Code Point (PCP) : a 3-bit field which refers to the IEEE 802.1p priority. It indicates the frame prioritylevel. Values are from 0 (best effort) to 7 (highest); 1 represents the lowest priority. These values can be usedto prioritize different classes of traffic (voice, video, data, etc.). See also Class of Service or CoS.

Canonical Format Indicator (CFI) : a 1-bit field. If the value of this field is 1, the MAC address is in non-canonical format. If the value is 0, the MAC address is in canonical format. It is always set to zero for Ethernetswitches. CFI is used for compatibility between Ethernet and Token Ring networks. If a frame received at anEthernet port has a CFI set to 1, then that frame should not be bridged to an untagged port.

VLAN Identifier (VID) : a 12-bit field specifying the VLAN to which the frame belongs. The hexadecimal valuesof 0x000 and 0xFFF are reserved. All other values may be used as VLAN identifiers, allowing up to 4,094VLANs. The reserved value 0x000 indicates that the frame does not belong to any VLAN; in this case, the802.1Q tag specifies only a priority and is referred to as a priority tag . On bridges, VLAN 1 (the default VLANID) is often reserved for a management VLAN; this is vendor-specific.

fromhttp://en.wikipedia.org/wiki/IEEE_802.1Q


In this 802.1Q tagged Ethernet frame, which block of bits, labeled A, B, C, and D, is used as VID?

A. AB. BC. CD. D



Tag Protocol Identifier (TPID) : a 16-bit field set to a value of 0x8100 in order to identify the frame as an IEEE802.1Q-tagged frame. This field is located at the same position as the EtherType/Length field in untaggedframes, and is thus used to distinguish the frame from untagged frames.

Priority Code Point (PCP) : a 3-bit field which refers to the IEEE 802.1p priority. It indicates the frame prioritylevel. Values are from 0 (best effort) to 7 (highest); 1 represents the lowest priority. These values can be usedto prioritize different classes of traffic (voice, video, data, etc.). See also Class of Service or CoS.

Canonical Format Indicator (CFI) : a 1-bit field. If the value of this field is 1, the MAC address is in non-canonical format. If the value is 0, the MAC address is in canonical format. It is always set to zero for Ethernetswitches. CFI is used for compatibility between Ethernet and Token Ring networks. If a frame received at anEthernet port has a CFI set to 1, then that frame should not be bridged to an untagged port.

VLAN Identifier (VID) : a 12-bit field specifying the VLAN to which the frame belongs. The hexadecimal valuesof 0x000 and 0xFFF are reserved. All other values may be used as VLAN identifiers, allowing up to 4,094VLANs. The reserved value 0x000 indicates that the frame does not belong to any VLAN; in this case, the802.1Q tag specifies only a priority and is referred to as a priority tag . On bridges, VLAN 1 (the default VLANID) is often reserved for a management VLAN; this is vendor-specific.

fromhttp://en.wikipedia.org/wiki/IEEE_802.1Q


In this IPv4 packet, which bits in the ToS byte are used for ECN?

A. bits 0, 1

B. bits 0, 1, 2C. bits 2, 3, 4D. bits 5, 6, 7



Seems they have it the other way round here. Matter of fact, ECN is only 2 bits, so only valid answer is A

fromhttp://en.wikipedia.org/wiki/Type_of_Service


In this IPv4 packet, which bits in the ToS byte are used for DSCP?

A. bits 0 to 5B. bits 2 to 7C. bits 3 to 7D. bits 5 to 7



Numbers are reversedm, but DSCP bits are 6 in total, so only answer B can be correct



In this IPv4 packet, which bits in the ToS byte are used for IP precedence?

A. bits 0 to 2B. bits 2 to 4C. bits 4 to 7D. bits 5 to 7



3 bits, so answer D must be correct


QUESTION 82Your client has a business requirement that mandates exact DTMF durations being passed end- to-end acrossan H.323 VoIP infrastructure. Which two DTMF relay methods meet the client requirement? (Choose two.)

A. Cisco RTPB. H.245 signalC. H.245 alphanumericD. RTP-NTEE. H.225 NotifyF. in-band voice



The "h245-signal" option relays a more accurate representation of a DTMF digit than the "h245-alphanumeric"option, in that tone duration information is included along with the dig it value .The "h245-alphanumeric" option simply relays DTMF tones as ASCII characters.There is no durationinformation associated with tones in this mode

RTP-NTE equals RFC2833 and duration information is kept.

Source

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/srnd/design/guide/multcme.html

QUESTION 83Cisco Unified Communications Manager generates different types of alarms to indicate system- or process-related problems. "Code Yellow" is one of these alarms. Which of these system or process exceptions willtrigger a Code Yellow alarm on Cisco Unified Communications Manager?

A. when a hard drive failsB. when there is a memory leakC. when the Cisco Unified Communications Manager application generates a core dumpD. when a database replication problem arisesE. when calls are throttled because of an unacceptably high delay in call handling



Call throttling allows Cisco Unified CallManager to automatically throttle (deny) new call attempts when itdetermines that various factors, such as heavy call activity, low CPU availability to Cisco Unified CallManager,routing loops, disk I/O limitations, disk fragmentation or other such events, could result in a potential delay todial tone (the interval users experience from going off hook until they receive dial tone).

fromhttp://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/5_1_3/ccmfeat/fsclthrt.html

QUESTION 84In which two circumstances would Cisco Unified Communications Manager accept inbound H.323 calls fromunknown IP hosts? (Choose two.)

A. when inbound H.323 calls are routed via gatekeeper-controlled trunksB. when inbound calls are routed via intercluster trunksC. after administrators have changed the Cisco Unified Communications Manager clusterwide service

parameter of "Accept Unknown TCP connection" to trueD. when inbound H.323 calls are routed via non-gatekeeper-controlled trunksE. when inbound calls are routed using H.323 fast startF. after administrators have changed the Cisco Unified Communications Manager clusterwide service

parameter of "Unknown Caller ID Flag" to true

Correct Answer: ACSection: (none)Explanation


QUESTION 85The Cisco UMR feature allows Cisco Unity to take outside caller messages while their Exchange Server isunavailable. Which two statements about Cisco UMR are incorrect? (Choose two.)

A. If the Cisco Unity primary Exchange Server goes offline, all subscribers hear the UMR conversation.B. Cisco Unity messages, deposited while the Message Store is down, will have different time stamps after the

Message Store returns to service and handles the message delivery.C. When Cisco Unity moves messages from Cisco UMR to the Exchange Server, all messages appear as new

even if they were listened to using the UMR conversation, thus also triggering MWIs.D. Cisco Unity does not light MWIs for messages that arrived during an outage and are in Cisco UMR.E. The Cisco UMR messages that Cisco Unity handled during an Exchange outage are stored in the local

directory at "C:\Commserver\UnityMTA". This path is hardcoded and cannot be changed after the CiscoUnity installation.

F. During an Exchange outage, messages to the unaddressed message distribution lists appear in Cisco UMRand can be accessed by all members of the list.

Correct Answer: EFSection: (none)Explanation


Specifically, the space available in the C:\Commserver\UnityMTA directory. This directory is controlled by aregistry settingDuring an Exchange outage messages to distribution lists, such as the Unaddressed Messages distribution list,appear in the UMR but are addressed to the distribution list and not to the members of the distribution list

fromhttp://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_tech_note09186a00800949fa.shtml

QUESTION 86Cisco Unity extends a number of schema object classes in Microsoft Active Directory during the schemaextension process. Which three object classes are extended by the Cisco Unity schema extension process?(Choose three.)

A. userB. computerC. domainD. organizational unitE. groupF. contact

Correct Answer: AEFSection: (none)Explanation



Using information that is provided in the Cisco IOS gatekeeper configuration and the show gatekeeper endpointoutput, how will the gatekeeper route the call when it receives an ARQ with a called number of 1000?

A. The call will be extended to the device with the H.323 ID of "cucm".B. The call will be extended to the device with the H.323 ID of "cme".C. The information that is provided is insufficient to answer the question. The output of show gatekeeper gw-

type-prefix is needed to determine the gateway selection decision of the gatekeeper.D. The call will be rejected by the gatekeeper.E. The information that is provided is insufficient to answer the question. The output of show gatekeeper zone

status for the bandwidth consumption level is needed to determine if the gatekeeper will admit the call.




What will be experienced by a PSTN caller when calling into this T1 PRI circuit?

A. The caller will hear a continuous ringback tone.B. The caller will hear a dial tone.C. The caller will hear a fast-busy tone.D. The caller will hear a slow-busy tone.E. The caller will not hear anything.



QUESTION 89Which set of SIP headers is mandatory in SIP requests?

A. Call-ID, Contact, User-Agent, RSeq, SDPB. Allow, Supported, Via, From, To, CSeqC. Via, From, To, Call-ID, CSeq, ContactD. Content-Type, Content-Length, Session-Expires, Via, FromE. Req URI, From, Via, To, CSeq



To, From, CSeq, Call-ID, Max-Forwards, and Via ; all of these header fields are mandatory in all SIP requests.These six header fields are the fundamental building blocks of a SIP message

fromhttp://tools.ietf.org/html/rfc3261#page-29

QUESTION 90Which statement regarding SIP requests or responses is correct?

A. SIP requests always expire after 120 seconds; this is known as the J-Timer in SIP RFC.B. Secure SIP requests using TLS cannot be interworked to non-TLS networks.C. SIP responses are always sent to the IP or FQDN in the "Via" header of an incoming request.D. The "Max-Forwards" header value is incremented as it passes through each SIP hop.E. Midcall SIP requests are always sent to the IP or FQDN in the "Contact" header of an incoming request.



Topic 2, Volume B

QUESTION 91Which statement about the offer/answer model of SDP is correct?

A. Offer/answer cannot be considered complete when it happens in an INVITE/18x exchange.B. PRACK message must not carry SDP, or else offer/answer will not work.C. Offer must be included in the initial INVITE; otherwise, offer/answer cannot complete.D. It is best to start a call without an offer and wait for an answer.E. ACK message can carry the SDP answer.



QUESTION 92Which two mechanisms can be used to detect SIP calls that are hung or stuck in an incomplete state? (Choosetwo.)

A. SDP time stamps and version numberB. PRACK (RSeq)C. session timerD. periodic hold/resumeE. OOD ReferF. RTP and RTCP inactivity monitoring



QUESTION 93Which statement correctly describes symmetric signaling in SIP?

A. SIP devices use the same listening port for all incoming SIP messages.B. SIP devices send and receive SIP messages at the same time.C. SIP devices send SIP traffic to the same IP address and port number of an upstream element.D. SIP devices use the same source port for SIP messages.E. SIP devices use the same port number for sending and receiving SIP messages.




The line that is shown in the exhibit appeared in the Cisco Unified Communications Manager trace for a SIP IPphone that deregisters frequently. What could be the reason for deregistration?

A. The phone was reset from the Cisco Unified Communications Manager Administration web page.B. The phone lost power momentarily and rebooted itself.C. Cisco Unified Communications Manager reset the TCP connection to the phone.D. The phone was restarted from the Cisco Unified Communications Manager Administration web page.E. The phone aborted the TCP connection.



QUESTION 95Which statement about the "Unknown Caller ID" service parameter in Cisco Unified Communications Managerconfiguration is true?

A. This parameter defines a numeric string to be displayed to the called party on inbound calls that arrived withno caller ID information.

B. This parameter designates a numeric string to be displayed to the called party for outbound calls withoutcaller ID information.

C. This parameter defines a numeric or a text string to be displayed to the called party for inbound calls thatarrived with no caller ID information.

D. This parameter designates a numeric or a text string to be displayed to the called party for outbound callswithout caller ID information.

E. This parameter defines a numeric string to be displayed to the called party on inbound and outbound callswith no caller ID information.



UnknownCallerId: The directory number to be displayed. Valid value is any numeric value representing ageneral number for your system (if you wish to provide caller ID functionality to called parties). Valid value is anyvalid telephone number.

QUESTION 96Which two media resources are not required by Cisco Unified Communications Manager for outbound early-offer support on SIP trunks? (Choose two.)

A. annunciatorB. software-based MTP in Cisco IOS gatewaysC. hardware-based MTPs in Cisco IOS gatewaysD. software-based MTPs using the Cisco IP Voice Media Streaming Application on Cisco MCSE. Cisco Unified Border Element with H.323-to-SIP interworking enabled



QUESTION 97There are several methods to transport DTMF digits between SIP endpoints. Which three methods aresupported by Cisco Unified Communications Manager? (Choose three.)

A. Unsolicited NotifyB. INFOC. KPMLD. RFC 2833 in-band signal tone/eventE. Cisco RTP-NTEF. H.245 alphanumeric

Correct Answer: ACDSection: (none)Explanation


RTP-NTE and H.245 are H.323 counterparts. KPML and RFC2833 are confirmed.

QUESTION 98According to the Cisco QoS SRND guide, cRTP is recommended on which link speed?

A. lower than or equal to 10 Mb/sB. lower than or equal to 384 kb/sC. lower than or equal to 1.544 Mb/sD. lower than or equal to 768 kb/sE. lower than or equal to 512 kb/s



QUESTION 99Which Cisco IOS CLI command can be used to identify the high jitter level of an RTP stream on a Cisco IOSvoice gateway?

A. show call active voice briefB. show voip rtp connectionsC. show voice dsp detailedD. show voice call summaryE. show policy-map interface



QUESTION 100According to RFC 3551, where default mappings between RTP payload type numbers and encodings aredefined, which RTP payload type corresponds to G.711 a-law encoding?

A. 0B. 1C. 8D. 13E. 18



fromhttp://tools.ietf.org/html/rfc5391#section-4.5.14

QUESTION 101According to RFC 3551, where default mappings between RTP payload type numbers and encodings aredefined, which RTP payload type corresponds to encoded packets that are triggered by silence on a call withvoice activity detection?

A. 0B. 13C. 15D. 18

Correct Answer: BSection: (none)

Explanation


QUESTION 102Which version of MGCP is used on a Cisco IOS MGCP gateway that is registered to Cisco UnifiedCommunications Manager?

A. 0.0B. 0.1C. 1.0D. 1.1E. It varies between Cisco Unified Communications Manager versions.



MGCP call-agent: none Initial protocol service is MGCP 0.1

fromCisco ios

QUESTION 103On a Cisco IOS MGCP gateway, which DTMF relay method uses MGCP NTFY messages to send digits toCisco Unified Communications Manager?

A. CiscoB. NSEC. NTE-CAD. NTE-GWE. out-of-band



voip Specifies Voice-over-IP calls.

voaal2 Specifies Voice-over-AAL2 calls (using Annex K type 3 packets).

all Configures Dual Tone Multifrequency (DTMF) relay to be used with all voicecodecs.

low-bit-rate

Configures DTMF relay to be used with only low-bit-rate voice codecs, suchas G.729.

Real-time Transport Protocol (RTP) digit events are encoded using a

cisco proprietary format similar to frame relay as described in the FRF.11specification. The events are transmitted in the same RTP stream as non-digit voice samples, using payload type 121.

nse RTP digit events are encoded using the format specified in RFC 2833,Section 3.0, and are transmitted in the same RTP stream as non-digit voicesamples, using the payload type that is configured using the mgcp tsepayload command.

out-of-band

Media gateway control protocol (MGCP) digit events are sent using NTFYmessages to the call agent (CA), which plays them on the remote GW usingRQNT messages with S: (signal playout request).

nte-gw RTP digit events are encoded using the format specified in RFC 2833,Section 3.0, and are transmitted in the same RTP stream as non-digit voicesamples. The payload type is negotiated by the GWs before use. Theconfigured value for payload type is presented as the preferred choice at thebeginning of the negotiation.

nte-ca Identical to the nte-gw keyword behavior except that the CA's localconnection options a: line is used to enable or disable DTMF relay.

fromhttp://www.cisco.com/en/US/docs/ios/12_2t/12_2t8/feature/guide/ftmgcpfx.html

QUESTION 104Which SCCP message is used to instruct to an SCCP IP phone the remote IP address and port number tosend RTP packets?

A. Station IP Port messageB. Station Open Receive Channel messageC. Station Start Media Transmission messageD. Station Call Information messageE. Station Open Logical Channel message



Unified CM instructs both phones to Start Media Transmission to each other's IP addresses. The phones areonce again connected via an RTP two-way audio stream.

fromhttp://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/5x/50moh.html

QUESTION 105An IP phone user just answered an incoming call by lifting the handset. Assuming that the IP phone usesSCCP, which SCCP message will Cisco Unified Communications Manager transmit to this called IP phoneimmediately after receiving notification about the off-hook event?

A. Station Media Port List message

B. Station Set Ringer messageC. Station Stop Tone messageD. Station Start Media Transmission messageE. Station Open Receive Channel message



Station Set Ringer (Off)

fromhttp://www.cisco.com/en/US/docs/voice_ip_comm/cata/186_188/2_15_ms/english/administration/guide/sccp/sccpaaph.pdf

QUESTION 106Which SCCP message is used by an IP phone to inform Cisco Unified Communications Manager about the IPaddress and port number to be used for an incoming RTP stream?

A. Station Capability Response messageB. Station IP Port messageC. Station Open Receive Channel ACK messageD. Station Media Reception ACK messageE. Station Start Media Transmission message



fromhttp://www.cisco.com/en/US/docs/voice_ip_comm/cata/186_188/2_15_ms/english/administration/guide/sccp/sccpaaph.pdf

QUESTION 107When an IP phone that is using SCCP places an active call on hold, which SCCP message will be transmittedfrom the phone to Cisco Unified Communications Manager?

A. Station On Hold messageB. Station Keypad Button message

C. Station Close Receive Channel messageD. Station Stop Media Transmission messageE. Station Softkey Event message



QUESTION 108What is the proper Cisco IOS CLI command to configure an analog FXS port to be controlled by Cisco UnifiedCommunications Manager using SCCP?

A. dial-peer voice 1 potsport 1/0service skinny

B. dial-peer voice 1 potsport 1/0service sccp

C. dial-peer voice 1 potsport 1/0service stcapp

D. dial-peer voice 1 potsport 1/0service sccpapp

E. dial-peer voice 1 potsport 1/0application sccp



from http://www.cisco.com/en/US/docs/ios/12_4t/12_4t2/ht1vg224.html


Listed in the exhibit are five attributes that a Cisco IOS router uses to select an inbound dial peer. Whichattribute order, from highest to lowest priority, is used by a Cisco IOS router for inbound dial-peer matching?

A. II, I, III, V, IVB. III, I, II, V, IVC. III, II, I, V, IVD. II, III, I, V, IVE. V, III, II, I, IVF. V, II, III, I, IV



1. Called number (DNIS) with the incoming called-number command2. Calling Number (ANI) with the answer-address command3. Calling Number (ANI) with the destination-pattern command4. Voice-port (associated with the incoming call setup request) with configured dial peer port (applicable for

inbound POTS call legs)5. If no match is found in the first four steps, then the default dial peer 0 (pid:0) command is used.

fromhttp://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml#topic3

QUESTION 110Which two attributes are not used by a Cisco IOS router in the inbound dial-peer selection process? (Choosetwo.)

A. default dial-peer 0B. called number with the destination-pattern command of each dial peerC. calling number with the destination-pattern command of each dial peerD. calling number with the answer-address command of each dial peerE. called number with the incoming called-number command of each dial peerF. called number with the answer-address command of each dial peerG. voice port that is associated with an incoming call

Correct Answer: BFSection: (none)Explanation






When an inbound call with a calling number of 1001 and a called number of 2112 arrives at a Cisco IOS routerwith these dial peers, what is the correct order of dial-peer matching, from highest to lowest priority?

A. II, III, I, IVB. II, IV, III, IC. III, IV, II, ID. III, II, IV, IE. II, III, IV, I







When an outbound call with a calling number of 2112 and a called number of 1001 is placed through a CiscoIOS router with these dial peers, what is the correct order of dial-peer matching, from highest to lowest priority?

A. I, II, III, IVB. II, I, III, IVC. II, I, IV, IIID. I, II, IV, IIIE. II, IV, I, III



The method a router uses to select an outbound dial peer depends on whether ISDN DID is configured in theinbound POTS dial peer. If DID is not configured in the inbound POTS dial peer, the router collects theincoming dialed string digit by digit. As soon as one dial peer is matched, the router immediately places the callusing the configured attributes in the matching dial peer.

fromhttp://www.cisco.com/en/US/docs/ios/12_3/vvf_c/dial_peer/dp_confg.html#wp1068055

QUESTION 113What is the default DTMF relay for Cisco Unity Express when integrated via SIP?

A. RTP-NTEB. SIP NotifyC. SIP INFOD. in-band audioE. SIP Subscribe/Notify



fromhttps://supportforums.cisco.com/thread/2083151


This Cisco Unified Communications Manager trace shows a SIP message that is sent by a SIP Cisco Unified IPPhone 7965 to Cisco Unified Communications Manager. Which of these regarding the content of this SIPmessage is correct?

A. phone registration message to the primary Cisco Unified Communications ManagerB. keepalive message to the primary Cisco Unified Communications ManagerC. phone registration message to the secondary Cisco Unified Communications Manager during a server

failoverD. keepalive message to the secondary Cisco Unified Communications ManagerE. phone registration message to the primary Cisco Unified Communications Manager during fallback




This Cisco Unified Communications Manager trace shows a SIP message that was sent by a SIP Cisco UnifiedIP Phone 7965 to Cisco Unified Communications Manager. Which of these about the content of this SIPmessage is correct?

A. phone registration message to the primary Cisco Unified Communications ManagerB. keepalive message to the primary Cisco Unified Communications ManagerC. phone registration message to the secondary Cisco Unified Communications Manager during a server

failoverD. keepalive message to the secondary Cisco Unified Communications ManagerE. phone registration message to the primary Cisco Unified Communications Manager during fallback



QUESTION 116A customer purchased 10,000 phone license units for a Cisco Unified Communications Manager cluster. Howmany phone license unit overdrafts are permitted in this Cisco Unified Communications Manager cluster?

A. 200B. 500C. 700D. 1000E. Phone license unit overdrafts are never permitted.



5% overdraft. 5% of 10.000 = 500


The error alert that is shown in the exhibit is seen in the "Event Viewer--Application Log" on Cisco UnifiedPresence. Which action will be performed by the Cisco LPM tool in response to the alert?

A. LPM will purge trace and core files until disk usage is below the configured low watermark.B. LPM will purge all trace files and core files.C. LPM will not do anything; administrators must manually remove excess files in the active partition.D. LPM will not do anything; administrators must manually remove excess files in the common partition.E. LPM will purge some of the trace and core files until 50 percent of the disk space is available.



This alert occurs when the percentage of used disk space in the log partition exceeds the configured high watermark. When this alert gets generated, LPM deletes files in the log partition (down to low water mark) to avoidrunning out of disk space.

fromhttp://www.cisco.com/en/US/docs/voice_ip_comm/cucm/service/7_0_1/rtmt_master/rtalsys.html#wp1013680


The log was captured for a Cisco Unified Presence client that is not able to perform desk phone control to aCisco IP phone. Which two of these could be the potential causes that are revealed by the log? (Choose two.)

A. The IP phone is not registered.B. The IP phone is not configured with "Allow Control of Device from CTI."C. The directory number of the IP phone is not configured with "Allow Control of Device from CTI."D. The Standard CTI Enabled group is not added to the Cisco Unified Presence user in Cisco Unified

Communications Manager.

E. The CTI gateway profile is not added to the user application profile in Cisco Unified Presence.F. The Cisco CTIManager service is not running on Cisco Unified Communications Manager.



QUESTION 119Which network port is not used by the Cisco Unified Presence client?

A. TCP port 143B. TCP port 2000C. TCP port 2748D. UDP port 69E. TCP port 443F. TCP port 389



143 - used by IMAP2748 - used by CTI Gateway69 - used by TFTP443 - used by HTTPS389 - used by LDAP

fromhttp://www.cisco.com/en/US/docs/voice_ip_comm/cupc/7_0/english/release/notes/ol15710.html#wp39407

QUESTION 120When using the Local Route Group feature in Cisco Unified Communications Manager, in which two levels canyou apply the called party transformation pattern? (Choose two.)

A. device poolB. gatewayC. route patternD. route groupE. route listF. service parameter

Correct Answer: ABSection: (none)Explanation


verified in CUCM


Which dial peer will the Cisco IOS voice gateway match if an incoming call with a calling number of 100 andcalled number of 101 arrives at this T1 PRI port?

A. dial-peer voice 2B. dial-peer voice 3C. dial-peer voice 5D. dial-peer voice 8E. dial-peer voice 0







Which two statements about this debug message that was captured from a Cisco IOS H.323 gateway arecorrect? (Choose two.)

A. The calling-party number is 2000.B. If this gateway is able to accept the call, it should respond with an H.225 call proceeding message.C. If this gateway is able to accept the call, it should respond with an H.225 setup ACK message.D. If this gateway is able to accept the call, it should respond with an H.225 connect ACK message.E. The called-party number is 2000.F. Neither gateway is allowed to begin RTP transmission until the H.225 connect message is sent.



QUESTION 124Which Cisco IOS command and configuration mode can be used to force a Cisco IOS voice gateway to useTCP as the transport protocol for SIP?

A. router(config)#sip transport tcpB. router(conf-voi-serv)#no sip transport udpC. router(conf-serv-sip)#no transport udpD. router(conf-serv-sip)#transport tcpE. router(config-sip-ua)#no transport udp



fromhttp://www.ciscopress.com/articles/article.asp?p=664148&seqNum=6


The exhibit shows how MOH Server A and MOH Server B are associated with Phone A and Phone B. If PhoneA presses the Hold softkey during an active call with Phone B using the G.711 mu-law codec, which twostatements are correct? (Choose two.)

A. MOH Server A will be used to play MOH toward Phone B.B. MOH Server B will be used to play MOH toward Phone B.C. Phone B will continue using G.711 as the codec to receive MOH.D. Phone B will use G.729 as the codec to receive MOH by default.E. Phone B will renegotiate the codec with the selected MOH server based on the region settings of both

parties.



QUESTION 126When calls are placed by certain Cisco Unified Communications Manager supplementary services, the LocalRoute Group feature will be bypassed. Which of these does not belong to the supplementary services?

A. Call BackB. Call ForwardC. Message Waiting IndicatorD. Mobility Follow MeE. Path Replacement




QUESTION 127Phone A, Phone B, and Phone C are configured to be in Device Pool A, Device Pool B, and Device Pool C,respectively. The Local Route Group feature was configured on Cisco Unified Communications Manager foreach device pool. Phone B has set CFA to Phone C; Phone C has set CFNA to a PSTN number. When PhoneA calls Phone B and if Phone C does not answer, which local route group will be used to route the call?

A. The local route group that is configured for the Phone A device pool will be used.B. The local route group that is configured for the Phone B device pool will be used.C. The local route group that is configured for the Phone C device pool will be used.D. All local route groups will be bypassed.E. Cisco Unified Communications Manager will disallow the forwarded call because it might cause routing

loops.




The exhibit shows the T.30 message exchanges that resulted in a single page fax call failure. Which T.30message sequence will result in a successful fax transmission?

A. PPS, EOP, PPR, PPS, EOP, NSF, DCNB. MPS, EOP, PPR, PPS, EOP, MCF, DCNC. PPS, EOP, RTP, PPS, EOP, NSF, DCND. PPS, EOP, PPR, PPS, EOP, MCF, DCNE. PPS, EOP, NSF, DCN




The debug outputs that are shown in the exhibit were collected at the terminating Cisco IOS gateway for a faxcall that failed. Which two of these could be the failure reasons? (Choose two.)

A. The fax originated from a third-party fax gateway.B. The fax originated from a Cisco gateway that is configured with a protocol-based Cisco fax relay.C. The fax originated from a Cisco gateway that is configured with a protocol-based fax pass- through.D. The fax originated from a Cisco gateway that is configured with an NTE-based fax pass- through.E. The fax originated from a gateway that is configured with an NSE-based fax relay.



QUESTION 130Which MGCP message is used to indicate fax switchover in a call agent-controlled T.38 fax relay?

A. CRCXB. NSEC. NTED. NTFYE. MDCX














The debug that is shown was captured on a Cisco Unified Communications Manager Express router with FXO

ports connecting to the PSTN. All incoming calls from the PSTN are directed to an IP phone operator for furtherprocessing. The IP phone operator has reported that the calling number and name are absent for all incomingPSTN calls. Which configuration will resolve this issue?

A.

B. 65

C.

D.

E. Exhibit AF. Exhibit BG. Exhibit CH. Exhibit D




Which two DTMF capabilities are advertised in this SIP INVITE message? (Choose two.)

A. in-band voiceB. RTP-NTEC. SIP KPMLD. SIP NotifyE. Cisco RTPF. RTP-NSE




Which three header fields do not change for the duration of this call using SIP? (Choose three.)

A. From tagB. To tagC. Contact:D. transaction ID in the ViheaderE. request URIF. Call-ID:

Correct Answer: ABFSection: (none)Explanation



Which data rate, in bits per second, would be negotiated for a T.38 fax call?

A. 33600B. 28800C. 14400D. 24000E. 19200



QUESTION 135Which three statements about a modem pass-through call are correct? (Choose three.)

A. Clear-channel codec is used to transport modem tones.B. G.711 mu-law codec is used to transport modem tones.C. VAD is disabled.D. VAD is enabled.E. NLP is disabled.F. NLP is enabled.

Correct Answer: BCESection: (none)Explanation


fromhttp://www.cisco.com/en/US/docs/ios/voice/fax/configuration/guide/vf_cfg_mdm_psthr.pdf


What is the correct duration, in milliseconds, of the DTMF digit that is received?

A. 30B. 40C. 60D. 65E. 101



QUESTION 137Which three services must be activated on Cisco Unified Presence in order for presence and instant messagingto be functional? (Choose three.)

A. Cisco Unified Presence SIP ProxyB. Cisco AXL Web ServiceC. Cisco Bulk Provisioning ServiceD. Cisco Unified Presence EngineE. Cisco Unified Presence Sync AgentF. Cisco Serviceability Reporter



QUESTION 138Which Cisco Unified Presence service parameter must be modified from the default value in order for presenceand instant messaging to be functional?

A. server nameB. server IP addressC. DNS domainD. SIP proxy domainE. enable presence



QUESTION 139What is required to back up Cisco Unified Presence configuration?

A. tape backup deviceB. USB hard diskC. FTP serverD. SFTP serverE. TFTP server




Which of the certificates that are shown must be uploaded to Cisco Unified Presence when integrating thecalendar with Exchange Server "email.cisco.com"?

A. DST Root CA X3 onlyB. Cisco SSCA onlyC. email.cisco.com onlyD. DST Root CA X3 and Cisco SSCAE. Cisco SSCA and email.cisco.com



QUESTION 141Which of these best describes the "Incoming ACL" configuration on Cisco Unified Presence?

A. permits incoming packets to Cisco Unified PresenceB. bypasses digest authenticationC. allows instant messages

D. allows incoming certificates to Cisco Unified PresenceE. filters incoming presence status requests



QUESTION 142Which Cisco tool can be used to capture packets on Cisco Unified Presence?

A. Cisco Unified Presence CLIB. System Troubleshooter on the Cisco Unified Presence web portalC. Cisco Unified RTMTD. Cisco Unified Presence "Cisco Unified Serviceability" web portalE. Cisco Unified Presence "Cisco Unified OS Administration" web portal




User "jdoe" was not able to download voice mail with his Cisco Unified Personal Communicator. Whichconfiguration change on the Voicemail Profile Configuration page on Cisco Unified Presence is most likely tosolve this problem?

A. Change the Name field to the IP address.B. Select the appropriate option in the Voice Messaging Pilot field.C. Select the appropriate option in the Primary Voicemail Server field.D. Select the appropriate option in the Primary Mailstore field.E. Check the "Make this the default Voicemail Profile for the system" check box.



QUESTION 144Which two Cisco Unified Contact Center Express system components do not support integration redundancy?(Choose two.)

A. CTI portsB. AXL serviceC. Cisco Unified CM Telephony triggerD. CSQE. dialog groupsF. HTTP trigger

Correct Answer: DFSection: (none)Explanation


QUESTION 145Which three statements about the Outbound Dialer solution on Cisco Unified Contact Center Express arecorrect? (Choose three.)

A. The Outbound Dialer can make a call as long as the CTI port is available.B. In a Cisco Unified Contact Center Express high-availability system, the Outbound Dialer would not be

functional if one of the database nodes is down.C. When the Outbound Dialer makes a call to an invalid number, the system disconnects the call automatically

and will not involve any agent.D. The Outbound Dialer cannot use the Cisco IP Phone Agent to make calls.E. When the Outbound Dialer selects an agent to take a call, the agent will be given a choice whether to

accept the call.

Correct Answer: BDESection: (none)Explanation


QUESTION 146Which two statements about the Agent Email feature on Cisco Unified Contact Center Express are correct?(Choose two.)

A. An email-capable agent can receive both an incoming call and email at the same time.B. This feature supports IMAPv4, POP3, and SMTP email protocols.C. All email routing rules are configured at the Cisco Unified Contact Center Express Administration web

interface.D. An agent can use either Cisco Agent Desktop or Cisco Agent Desktop--Browser Edition to answer the

email.E. To make an agent email capable, assign the agent to an email CSQ.



QUESTION 147You have discovered that the Cisco Unified CM Telephony subsystem is in "PARTIAL_SERVICE" on a CiscoUnified Contact Center Express server. Which two misconfigurations could lead to this service state? (Choosetwo.)

A. The Cisco Unified Communications Manager JTAPI user has invalid login credentials.B. Not all CTI ports and CTI route points are associated with the Cisco Unified Communications Manager

JTAPI user.C. The hostname/IP address for Cisco Unified Communications Manager is incorrect.D. Not all agent phones are associated with the Cisco Unified Communications Manager JTAPI user.E. An invalid Cisco Unified Contact Center Express script is used by one of the applications.



QUESTION 148After logging into Cisco Agent Desktop, a Cisco Unified Contact Center Express agent could not go into readystate. Which two reasons could lead to this failure? (Choose two.)

A. The agent has not been assigned to any CSQ.B. The agent IP phone lost network connectivity.C. The agent has entered incorrect login credentials.D. The agent supervisor has not logged in.E. The agent IP phone has not been associated with the agent user in Cisco Unified Communications

Manager.



QUESTION 149Which of these best describes packetization delay in a VoIP network?

A. the time that is taken by the DSP to compress a block of PCM samplesB. the time that is taken by the compression algorithm to correctly process sample block NC. the time that is taken to fill a packet payload with encoded/compressed speechD. the time that is required to clock a voice frame onto the network interfaceE. the time that is taken to queue a voice frame for transmission on the network connection



QUESTION 150Which of these best describes encoder delay in a VoIP network?

A. the time that is taken by the compression algorithm to correctly process sample block NB. the time that is taken by the DSP to compress a block of PCM samplesC. the time that is taken to fill a packet payload with encoded/compressed speechD. the time that is required to clock a voice frame onto the network interfaceE. the time that is taken to queue a voice frame for transmission on the network connection



QUESTION 151Which delay in a VoIP network is also known as accumulation delay?

A. coder delayB. network switching delayC. queuing delayD. packetization delayE. dejitter delay



QUESTION 152Which statement about coder delay in a VoIP network is correct?

A. Coder delay is the time that is taken to fill a packet payload with encoded/compressed speech.B. Coder delay is also known as algorithmic delay.C. Coder delay transforms a variable delay into a fixed delay.D. Coder delay varies with the voice coder that is used and the processor speed.E. Coder delay compensates for network switching delay.



QUESTION 153Which Cisco IOS command is used to define the size of the jitter buffer on Cisco IOS VoIP gateways?

A. jitter-bufferB. expect-factor

C. acc-qosD. playout-delayE. dejitter-buffer



QUESTION 154Which three Cisco IOS commands can be used to verify configured playout delay values on Cisco VoIPgateways? (Choose three.)

A. show voice call summaryB. show call active voiceC. show dial-peer voice tag number for dial peerD. show voice port voice interface numberE. show voice dsp detailF. show voice accounting method

Correct Answer: BCDSection: (none)Explanation


QUESTION 155Which two characteristics about traffic shaping on Cisco IOS VoIP gateways are incorrect? (Choose two.)

A. Traffic shaping propagates burst.B. Traffic shaping buffers and queues excess packets above the committed rates.C. Traffic shaping token values are configured in bits per second.D. Traffic shaping is applicable to both inbound and outbound traffic.E. FRTS and generic traffic shaping are two ways of implementing traffic shaping.F. Traffic shaping could introduce delays becaus of deep queues.

Correct Answer: ADSection: (none)Explanation


QUESTION 156Which two characteristics about traffic policing on Cisco IOS VoIP gateways are correct? (Choose two.)

A. Traffic policing buffers and re-marks excess packets above the committed rates.B. Traffic policing propagates burst.C. Traffic policing token values are configured in bits per second.D. Traffic policing is applicable to both inbound and outbound traffic.E. Traffic policing is an inbound-only concept.

F. Traffic policing could introduce delays because of deep queues.



QUESTION 157CRTP belongs to which Cisco quality of service feature?

A. classificationB. congestion managementC. congestion avoidanceD. shaping and policingE. link efficiency mechanisms




Which statement about the QoS configuration for interface GigabitEthernet 1/0/1 on the Cisco Catalyst 3750Series Switch is correct?

A. Egress shaping is enabled with queue 1 being shaped to 25 percent of the available bandwidth.B. Egress sharing is enabled for all four queues; each queue is allocated 25 percent of the available

bandwidth.C. Egress shaping is disabled.D. Egress shaping is enabled with queue 1 being shaped to 4 percent of the available bandwidth.E. Egress shaping is enabled with queue 4 being shaped to 4 percent of the available bandwidth.




What is the correct expansion of the srr-queue abbreviation that is shown in the Cisco IOS command of theCatalyst 3750 Series Switch?

A. shared round-robin queueB. serviced round-robin queueC. shaped round-robin queueD. special round-robin queueE. serial round-robin queue



QUESTION 160Which statement about the Cisco Unity Connection message quota enforcement policies when a mailbox hasexceeded the send/receive quota is incorrect?

A. The user is unable to send messages.B. Cisco Unity Connection will automatically purge all deleted messages in the user mailbox.C. The user hears a warning that the message cannot be sent.D. Unidentified callers are not allowed to leave messages for the user.E. Messages from other users generate nondelivery receipts to the senders.



QUESTION 161What is the default mailbox size that triggers disablement of sending and receiving voice messages for a CiscoUnity Connection user?

A. 2 MBB. 4 MBC. 10 MBD. 14 MBE. 20 MB



Quota Level Mailbox SizeThat TriggersQuota Action

Action When Quota IsReached

Recording Time in Minutes Before Quota Is Reached

G.711Mu-Law

G.711 A-Law

G.726 32 Kbps

PCM 8 kHz

G.729a

Warning 12 megabytes The user is warned that themailbox is reaching themaximum size allowed.

25 25 50 50 200

Send 13 megabytes The user is prevented fromsending any more voicemessages.

27 27 54 54 217

Send/Receive

14 megabytes The user is prevented fromsending or receiving anymore voice messages.

31 31 61 61 246

fromhttp://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/administration/guide/7xcucsag190.html

QUESTION 162Which two statements about system broadcast messages on Cisco Unity Connection are correct? (Choosetwo.)

A. Users can fast-forward a system broadcast message.B. Users can save a system broadcast message.C. Users must listen to a system broadcast message in its entirety before they are allowed to hear new and

saved messages or to change setup options.D. A system broadcast message that has been listened to in its entirety will still be played again the next time

that the user logs in, but the user will be offered an option to skip it.E. If a user hangs up before playing the entire system broadcast message, the message plays again the next

time that the user logs in, as long as the message is still active.F. Users can forward a system broadcast message.



•System broadcast messages are played immediately after users log on to Cisco Unity Connection by phone—even before they hear message counts for new and saved messages. After logging on, users hear how manysystem broadcast messages they have and Connection begins playing them.

•For each system broadcast message, the sender specifies how long Connection broadcasts the message. Thesender can specify that a system broadcast message is "active" for a day, a week, a month—even indefinitely.A user hears the system broadcast message the first time that he or she logs on to Connection during theperiod that the message is active.

•Users must listen to a system broadcast message in its entirety before Connection allows them to hear newand saved messages or to change setup options. Users cannot fast-forward or skip a system broadcastmessage.

•If a user hangs up before playing the entire system broadcast message, the message plays again the nexttime that the user logs on to Connection by phone (assuming that the message is still active).

•When a user has finished playing a system broadcast message, the message can either be replayed orpermanently deleted. Users cannot respond to, forward, or save system broadcast messages.

•Users can receive an unlimited number of system broadcast messages.

•Users receive system broadcast messages even when they exceed their mailbox size limits and are no longerable to receive other messages. Because of the way that the messages are stored on the Connection server,they are not included in the total mailbox size for each user.

•New users hear all active system broadcast messages immediately after they enroll as Connection users.

•By design, system broadcast messages do not trigger message waiting indicators (MWIs) on user phones.They also do not trigger message notifications for alternative devices, such as a pager or another phone.

•Users hear broadcast messages only when listening to messages by phone. Users do not receive systembroadcast messages when listening to messages in the Cisco Unity Inbox, an RSS reader, IMAP clients, CiscoUnified Personal Communicator, or Cisco Unified Messaging with IBM Lotus Sametime.

•Connection does not respond to voice commands while playing broadcast messages. When using the voice-recognition input style, users will need to use key presses to either replay or delete the broadcast message.


QUESTION 163Which two statements about system broadcast messages on Cisco Unity Connection are correct? (Choosetwo.)

A. Users receive system broadcast messages even when they exceed their mailbox size limits and are nolonger able to receive other messages.

B. System broadcast messages trigger MWIs on user phones but do not trigger MWIs on alternate devicessuch as a pager.

C. Users hear broadcast messages only when listening to messages by phone.D. A system broadcast message that has been listened to in its entirety will still be played again the next time

that the user logs in, but the user will be offered an option to skip it.E. Users can only receive a limited number of system broadcast messages that are defined by the Cisco Unity

Connection Broadcast Message Administrator.F. Users can respond to a system broadcast message.



•System broadcast messages are played immediately after users log on to Cisco Unity Connection by phone—even before they hear message counts for new and saved messages. After logging on, users hear how manysystem broadcast messages they have and Connection begins playing them.

•For each system broadcast message, the sender specifies how long Connection broadcasts the message. The

sender can specify that a system broadcast message is "active" for a day, a week, a month—even indefinitely.A user hears the system broadcast message the first time that he or she logs on to Connection during theperiod that the message is active.

•Users must listen to a system broadcast message in its entirety before Connection allows them to hear newand saved messages or to change setup options. Users cannot fast-forward or skip a system broadcastmessage.

•If a user hangs up before playing the entire system broadcast message, the message plays again the nexttime that the user logs on to Connection by phone (assuming that the message is still active).

•When a user has finished playing a system broadcast message, the message can either be replayed orpermanently deleted. Users cannot respond to, forward, or save system broadcast messages.

•Users can receive an unlimited number of system broadcast messages.

•Users receive system broadcast messages even when they exceed their mailbox size limits and are no longerable to receive other messages. Because of the way that the messages are stored on the Connection server,they are not included in the total mailbox size for each user.

•New users hear all active system broadcast messages immediately after they enroll as Connection users.

•By design, system broadcast messages do not trigger message waiting indicators (MWIs) on user phones.They also do not trigger message notifications for alternative devices, such as a pager or another phone.

•Users hear broadcast messages only when listening to messages by phone. Users do not receive systembroadcast messages when listening to messages in the Cisco Unity Inbox, an RSS reader, IMAP clients, CiscoUnified Personal Communicator, or Cisco Unified Messaging with IBM Lotus Sametime.

•Connection does not respond to voice commands while playing broadcast messages. When using the voice-recognition input style, users will need to use key presses to either replay or delete the broadcast message.


QUESTION 164What is the maximum number of days for Cisco Unity Connection to retain expired system broadcastmessages?

A. 1 dayB. 5 daysC. 10 daysD. 30 daysE. 60 days



Retention Period—Indicates how long Connection retains expired system broadcast messages on the server.By default, Connection purges the WAV file and any data associated with a message 30 days after its end dateand time. To change the retention period for expired broadcast messages, enter a number from 1 to 60 days.


QUESTION 165What is the default maximum recording length that is allowed for system broadcast messages on a Cisco UnityConnection server?

A. 5 minutesB. 10 minutesC. 15 minutesD. 20 minutesE. 30 minutes



Maximum Recording Length—Indicates the maximum length allowed for system broadcast messages. Bydefault, senders can record messages up to 300,000 milliseconds (5 minutes) in length. To change themaximum recording length, enter a number from 60,000 (1 minute) to 36,000,000 (60 minutes) milliseconds.


QUESTION 166What is the maximum recording length that is allowed for system broadcast messages on a Cisco UnityConnection server?

A. 5 minutesB. 10 minutesC. 15 minutesD. 30 minutesE. 60 minutes



Maximum Recording Length—Indicates the maximum length allowed for system broadcast messages. Bydefault, senders can record messages up to 300,000 milliseconds (5 minutes) in length. To change themaximum recording length, enter a number from 60,000 (1 minute) to 36,000,000 (60 minutes) milliseconds.


QUESTION 167Which of these is not a valid VPIM message addressing option that is provided by Cisco Unity Connection toindividuals on a remote voice messaging system?

A. blind addressingB. Cisco Unity Connection directoryC. implicit addressingD. private distribution list

E. system distribution list



QUESTION 168Which three call handlers are predefined on Cisco Unity Connection? (Choose three.)

A. goodbyeB. holiday greetingC. internalD. opening greetingE. operatorF. closedG. standard



OpeningGreeting

Acts as an automated attendant, playing the greeting that callers first hear when they callyour organization, and performing the actions you specify. The Opening Greeting CallRouting rule transfers all incoming calls to the Opening Greeting call handler.

By default, the Opening Greeting call handler allows callers to press * to reach the Sign-inconversation, or press # to reach the Operator call handler. Messages left in the OpeningGreeting call handler are sent to the Undeliverable Messages distribution list.

Operator Calls are routed to this call handler when callers press "0" or do not press any key (thedefault setting). You can set up the Operator call handler so that callers can leave amessage or be transferred to a live operator.

By default, the Operator call handler allows callers to press * to reach the Sign-inconversation, or press # to reach the Opening Greeting call handler. Messages left in theOperator call handler are sent to the mailbox for the Operator user.

Goodbye Plays a brief goodbye message and then hangs up if there is no caller input.

By default, the Goodbye call handler allows callers to press * to reach the Sign-inconversation, or press # to reach the Opening Greeting call handler. If you change theAfter Greeting action from Hang Up to Take Message, then messages left in the Goodbyecall handler are sent to the Undeliverable Messages distribution list.

fromhttp://www.cisco.com/en/US/docs/voice_ip_comm/connection/1x/administration/guide/acm030.html

QUESTION 169Which greeting type is not a valid call handler on Cisco Unity Connection?

A. busy

B. closedC. externalD. holidayE. standard



Standard Plays at all times unless overridden by another greeting. You cannot disable the standard greeting.

Closed Plays during the closed (nonbusiness) hours defined for the active schedule. A closed greetingoverrides the standard greeting, and thus limits the standard greeting to the open hours defined forthe active schedule.

Holiday Plays during the specific dates and times specified in the schedule of holidays associated with theactive schedule. A holiday greeting overrides the standard and closed greetings.

Internal Plays to internal callers only. It can provide information that only coworkers need to know. (Forexample, "I will be in the lab all afternoon.") An internal greeting overrides the standard, closed, andholiday greetings.

Not all phone system integrations provide the support necessary for an internal greeting.

Busy Plays when the extension is busy. (For example, "All of our operators are with other customers.") Abusy greeting overrides the standard, closed, internal, and holiday greetings.

Not all phone system integrations provide the support necessary for a busy greeting.

Alternate Can be used for a variety of special situations, such as vacations or a leave of absence. (Forexample, "I will be out of the office until....") An alternate greeting overrides all other greetings.

Error Plays if the caller enters invalid digits. This can happen if the digits do not match an extension, theextension is not found in the search scope, or the caller is otherwise restricted from dialing thedigits. You cannot disable the error greeting.

The system default error recording is, "I did not recognize that as a valid entry." By default, after theerror greeting plays, Connection replays the greeting that was playing when the caller entered theinvalid digits.

fromhttp://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/administration/guide/7xcucsag060.html#wp1052414

QUESTION 170Which two call handler greeting types on Cisco Unity Connection are overridden by the holiday greeting?(Choose two.)

A. alternateB. busyC. closedD. errorE. internalF. standard





Holiday Plays during the specific dates and times specified in the schedule of holidays associated with theactive schedule. A holiday greeting overrides the standard and close d greetings.









QUESTION 171Which three call handler greeting types on Cisco Unity Connection are overridden by the internal greeting?(Choose three.)

A. alternateB. busyC. closedD. errorE. holidayF. standard

Correct Answer: CEFSection: (none)Explanation





Internal Plays to internal callers only. It can provide information that only coworkers need to know. (Forexample, "I will be in the lab all afternoon.") An internal greeting overrides the standard, closed ,and holiday greetings.








QUESTION 172Which two call handler greeting types on Cisco Unity Connection cannot be disabled? (Choose two.)

A. alternateB. busyC. closedD. errorE. holidayF. standard

Correct Answer: DFSection: (none)Explanation


Standard Plays at all times unless overridden by another greeting.You cannot disable the standardgreeting .











QUESTION 173Which three options are not valid application types on Cisco Unified Contact Center Express? (Choose three.)

A. alternate applicationB. busyC. Cisco script applicationD. Cisco Unified CM TelephonyE. Ring No AnswerF. standard application

Correct Answer: ADFSection: (none)Explanation


QUESTION 174What is the correct variable type for the "CSQ" variable in the "icd.aef" script on Cisco Unified Contact CenterExpress?

A. stringB. userC. queue

D. documentE. Boolean



QUESTION 175What is the correct variable type for the "DelayWhileQueued" variable in the "icd.aef" script on Cisco UnifiedContact Center Express?

A. stringB. userC. numberD. integerE. Boolean



QUESTION 176Which statement about embedded Tcl scripts for B-ACD on Cisco Unified Communications Manager Expressis correct?

A. The Tcl scripts that are required for B-ACD services, along with the default audio files, must be available onthe router flash memory.

B. The Tcl scripts and the default audio files for B-ACD services are embedded natively in the Cisco IOSSoftware, eliminating the requirement to download these files to the router flash memory.

C. The Tcl scripts that are required for B-ACD services are embedded natively in the Cisco IOS Software;however, the default audio files must still be downloaded to the router flash memory.

D. The default audio files are embedded natively in the B-ACD Tcl scripts.E. The Tcl scripts and the default audio files for B-ACD services must be available on a TFTP server other

than the router itself



QUESTION 177Which of these is not a mandatory configuration component to enable B-ACD service on Cisco UnifiedCommunications Manager Express?

A. an automated attendant Tcl script that handles the welcome prompt and menu choicesB. a call-queue Tcl script that manages call routing and the queuing behavior numberC. an ephone hunt group to receive calls from the call-queue service

D. incoming dial peers for automated attendant pilot numbersE. Cisco Unity Express for voice mail to receive undelivered B-ACD calls



QUESTION 178On Cisco Unified Communications Manager Express with a B-ACD application that is provisioned for four huntgroups--aa-hunt1, aa-hunt2, aa-hunt3, and aa-hunt4--which hunt group will be chosen when a caller dials 0?

A. aa-hunt0B. aa-hunt1C. aa-hunt2D. aa-hunt3E. aa-hunt4



QUESTION 179What is the maximum number of calls that are allowed in each ephone hunt group call queue that is used byCisco Unified Communications Manager Express B-ACD?

A. 10B. 15C. 20D. 25E. 30



QUESTION 180What is the default number of calls that are allowed in each ephone hunt group call queue that is used by CiscoUnified Communications Manager Express B-ACD?

A. 5B. 10C. 15D. 20E. 30

Correct Answer: B

Section: (none)Explanation


QUESTION 181What is the maximum number of ephone hunt groups that can be used with a call-queue service by CiscoUnified Communications Manager Express B-ACD?

A. 3B. 5C. 10D. 15E. 20




What is the pilot number for the ephone hunt group that is configured on this Cisco Unified CommunicationsManager Express with B-ACD?

A. 1B. 30C. 3000D. 3333E. 5553000




Which statement about the B-ACD configuration on Cisco Unified Communications Manager Express iscorrect?

A. B-ACD will wait 20 seconds between retries to connect to an ephone hunt group pilot number.B. The B-ACD automated attendant script will play the "_bacd_welcome.au" file as soon as an incoming call is

answered.C. The caller is able to dial extension numbers when selecting menu option 3.D. Calls are answered and routed to a call queue immediately without invoking any interactive menu.E. The maximum number of calls that are waiting in the B-ACD queue is 20.



QUESTION 184

Which attribute is not associable with a device profile on Cisco Unified Communications Manager?

A. User Hold MOH Audio SourceB. phone button templateC. softkey templateD. directory URLE. expansion module information



QUESTION 185Which two attributes are associable with a device profile on Cisco Unified Communications Manager? (Choosetwo.)

A. MLPP informationB. Network Hold MOH Audio SourceC. privacyD. directory URLE. authentication service URL



QUESTION 186The Cisco Dialed Number Analyzer service belongs to which feature service group on Cisco UnifiedCommunications Manager?

A. Database and Admin ServicesB. Performance and Monitoring ServicesC. CM ServicesD. CTI ServicesE. Voice Quality Reporter Services



QUESTION 187The Cisco AXL Web Service belongs to which feature service group on Cisco Unified CommunicationsManager?

A. Database and Admin ServicesB. Performance and Monitoring Services

C. CM ServicesD. CTI ServicesE. Voice Quality Reporter Services



QUESTION 188The Cisco Unified Communications Manager Assistant service belongs to which feature service group on CiscoUnified Communications Manager?

A. Database and Admin ServicesB. Performance and Monitoring ServicesC. CM ServicesD. CTI ServicesE. Voice Quality Reporter Services



QUESTION 189Which two of these are valid modes of operation for the Cisco Unified Communications Manager Assistantfeature? (Choose two.)

A. forwarded line supportB. pickup line supportC. proxy line supportD. hybrid line supportE. shared line supportF. dual line support



QUESTION 190Which three features override the DND setting on an SCCP-controlled IP phone on Cisco UnifiedCommunications Manager? (Choose three.)

A. park reversion for remotely parked callsB. callback--terminating sideC. MLPPD. hold reversion

E. park reversion for locally parked callsF. remotely placed pickup request

Correct Answer: CDESection: (none)Explanation


QUESTION 191Which two features do not override the DND setting on an SCCP-controlled IP phone on Cisco UnifiedCommunications Manager? (Choose two.)

A. park reversion for remotely parked callsB. MLPPC. callback--terminating sideD. hold reversionE. intercomF. park reversion for locally parked calls



QUESTION 192Which statement about whisper intercom implementation on Cisco Unified Communications Manager iscorrect?

A. Only one-way audio exists from the calling to the called party.B. The speaker volume on the called phone will be reduced automatically to avoid disturbance to other users

nearby.C. The called party auto-answers the call in headset mode.D. Only one-way audio exists from the called to the calling party.E. Whisper Intercom is visual only, there is no audio.


Explanation/Reference:Explanation:When a phone user dials a whisper intercom line, the called phone automatically answers using speakerphonemode, providing a one-way voice path from the caller to the called party, regardless of whether the called partyis busy or idle.

Unlike the standard intercom feature, this feature allows an intercom call to a busy extension. The calling partycan only be heard by the recipient. The original caller on the receiving phone does not hear the whisper page.The phone receiving a whisper page displays the extension and name of the party initiating the whisper pageand Cisco Unified CME plays a zipzip tone before the called party hears the caller's voice. If the called partywants to speak to the caller, the called party selects the intercom line button on their phone. The lamp forintercom buttons are colored amber to indicate one-way audio for whisper intercom and green to indicate two-way audio for standard intercom.

You must configure a whisper intercom directory number for each phone that requires the Whisper Intercomfeature. A whisper intercom directory number can place calls only to another whisper intercom directorynumber. Calls between a whisper intercom directory number and a standard directory number or intercomdirectory number are rejected with a busy tone.

This feature is supported in Cisco Unified CME 7.1 and later versions

fromhttp://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeinter.html#wp1019638


Which two statements about the "Operational VLAN Id" parameter on the Cisco IP phone NetworkConfiguration menu are true? (Choose two.)

A. This parameter can be configured from the Cisco Unified Communications Manager Web AdministrationPhone Configuration page.

B. This parameter can be manually administered from the phone, as long as the Settings menu of the phone isunlocked.

C. This parameter is learned from the connected switch port.D. This parameter cannot be locally administered from the phone.E. This parameter can be configured by establishing an HTTP session to the IP address of the phone.

Correct Answer: CDSection: (none)Explanation



Which two statements about the "Admin. VLAN Id" parameter on the Cisco IP phone Network Configurationmenu are true? (Choose two.)

A. This parameter can be configured from the Cisco Unified Communications Manager Web AdministrationPhone Configuration page.

B. This parameter can be manually administered from the phone, as long as the Settings menu of the phone isunlocked.

C. This parameter is not learned from the connected switch port.D. This parameter cannot be locally administered from the phone.E. This parameter can be configured by establishing an HTTP session to the IP address of the phone.



QUESTION 195Which two strings are valid route patterns on Cisco Unified Communications Manager? (Choose two.)

A. 123+B. 123$C. 123/D. 123%E. 123DF. 123T



Placeholder Explanation Examples

Digit A digit that is interpreted as its literal value. 0 matches 0, 1 matches 1, 2 matches2, and so forth

X Any single digit in the range 0 through 9. X matches 0, 1, 2, 3, 4, 5, 6, 7, 8, or 9

[m-n] Any single digit in the range m through n. [4-9] matches 4, 5, 6, 7, 8, or 9

[^m-n] Any single digit outside of the range m throughn.

[^4-9] matches 0, 1, 2, 3, #, or *

! One or more digits in the range 0 through 9.This placeholder can be very useful forvariable length Dial Plans. The call is notrouted until the # key is pressed or the inter-digit timeout expires (default 10 seconds).

9! matches 91, 911, or 912342, butnot 9 only, as ! requires at least onedigit for a match.

? Same as ! except that it can match zero ormore digits.

9? matches 9, 91, 911, or 912342

+ One or more of the previous digits or aplaceholder.

9+ matches 99, 999, or 99999X+ is equivalent to 9! and 9X?

. A portion of the number that can be strippedafter a pattern is matched.

Match Pattern: 9.8XXXDiscard digits instruction (DDI): PreDotIf the user dials 98111, the PreDotDDI is applied to the 9.8XXX routepattern, which strips the “9” from thedialed digits and sends only the 8111to the PBX.

@ References the North American NumberingPlan (NANP), which is actually a macro thatcontains about 300 individual patterns. Youcan use this placeholder to apply filteringrules, because an @ makes “Area Code” and“Service Code” available to filter checks.Additional information on the tags is in the NANP Tags section of this document.

[2-9]11 matches 211, 311, 411, 511,611, 711, 811, or 911[2-9]XX XXXX matches 7-digit localnumbers[2-9]XX [2-9]XX XXXX matches 10-digit local numbers1 [2-9]XX [2-9]XX XXXX matches 11-digit long distance numbers011 3[0-469] ! matches internationalnumbers

fromhttp://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a00800949f0.shtml


All incoming calls to the Cisco Unified Communications Manager Express B-ACD are disconnectedimmediately. What is the reason for the failure?

A. The wrong Tcl is associated with POTS dial-peer 30. The correct Tcl script should be "app-b- acd".B. The drop-through option is in conflict with the welcome prompt.C. The param aa-pilot should be 3000 instead of 5553000.D. The mandatory command param voice-mail is missing.E. The number that is specified in param number-of-hunt-grps should be more than 1.




CCIE Voice 350-030 - GRATIS EXAM...Refer to the exhibit. 4 Traffic flows from the IP phone that is...

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