CCIE Voice 350-030 Q1.What 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. Ans: D, E Q2.Which 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. Ans: A, E Q3.Which two statements apply to the partitions function in Cisco Unified Communications Manager?(Choose two.) A. When a directory number or route pattern is placed into a certain partition, this creates a rule for who can call that device or route list B. A partition is a logical grouping of directory numbers and route patterns that have similar reachability characteristics. 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. Ans: A, B Q4.Which 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 the multicast 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. Ans: A, D, F

http://www.cisco.com/en/US/docs/switches/datacenter/nexus5000/sw/configuration/guide/cli/IGMPS nooping.html

Q5.Which three options are valid SCCP call states sent to an IP phone? A. Ring Off B. On Hook C. Call Transmit D. Connected E. Disconnected F. In Use Remotely Ans: B, D, F

The call states sent from Cisco Call Manager and understood by the SCCP endpoints:

1Off Hook 2On Hook 3Ring Out 4Ring In 5Connected 6Busy 7Line In Use 8Hold 9Call Waiting 10Call Transfer 11Call Park 12Call Proceed 13In Use Remotely 14Invalid Number

Q6.Which 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. Ans: A, D, E

Q7.Which two of the following are functions of DHCP snooping? (Choose two.) A. relies on already discovered trusted and untrusted ports B. dynamic ARP inspection C. defines trusted and untrusted ports D. uses existing binding tables E. builds a binding table F. automatically builds ACLs Ans: C, E Rate-limits DHCP traffic from trusted and untrusted sources. Builds and maintains the DHCP snooping binding database, which contains information about untrusted hosts with leased IP addresses.

Q8. Refer to Exhibit.

Traffic flows from the IP phone that is connected to SW1 to the IP phone on SW2. If the trust boundary has been extended to the IP phone on SW1, in what two places will traffic be marked and classified so that the proper QoS settings may be carried through the network? (Choose two.) A. IP phone attached to SW1 B. SW1 ingress port C. R1 ingress port D. SW1 egress port E. R1 egress port Ans: B, C

Q9. Refer to Exhibit.

Which gatekeeper mechanism prevents the gatekeeper from using all the resources on either gateway 1 or gateway 2 when sending calls to zones SE and NW? A. bandwidth remote B. resource availability indicator C. bandwidth total D. bandwidth zone E. Irq immediate advance F. ras timeout brq Ans: B
Resource availability indicators are particularly useful for gatekeepers that do not load share conferences over the available blades, but direct all conferences to a single blade. When that blade is full it informs the gatekeeper which can start to fill the next blade. When conferences finish and people disconnect, the blade sends an indication that it can accept new conferences again.

Q10. When implementing a Cisco Unified Communications Manager solution over an MPLS WAN, which two rules must 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. Ans: B, D

Q11.How 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 the OGW to send an NSF message to the TGW, instructing it to switch to the G.711 codec with echo chancellor and VAD disabled. Ans: B (TGW: Terminating Gateway)

Q12. Refer to the exhibit.

Explanation: G729=8kbit. 8x2=16. Interzone bandwidth = 96. 96/16=6

How many simultaneous G.729 calls can be established between sites SJ and RTP ? A. 4 B. 5 C. 6 D. 8 E. 12 Ans: C

Q13. Company Alpha has a central office and a branch office that utilize a central call processing topology. Calls between 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, which two of the following are possible solutions? (Choose two.) A. a software conference bridge that is configured in Cisco Unified Communications Manager B. a software conference bridge that is configured in Cisco Unified Communications Manager and a HW transcoder. C. a hardware conference bridge D. a hardware transcoder and a hardware conference bridge E. No extra configurations required-phones automatically negotiate using the lowest common denominator codec (G.729) Answer: B,C Software resources are not capable of transcoding, always Hardware required for this .

Q14. Refer to Exhibit

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 by extension, 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 drop it 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 existing Menu 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 step and 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. Ans: C Q15. Which two 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 Communications Manager. 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. Ans: B, D

Troubleshooting steps involved: 1. Refer to the Cisco Unified CCX trace files to determine what did not initialize. 2 Verify that all CTI ports and CTI route points are associated with the JTAPI user in Cisco Unified CM. 3. Verify that the Cisco Unified CM and JTAPI configuration IP addresses match. 4. Verify that the Cisco Unified CM JTAPI user has control of all the CTI ports and CTI route points. 5. Verify that the application file was uploaded to the repository using the Repository Manager.

Q16. Which three of these are mandatory sub-commands of the call-manager-fallback command and will help an IP phone register to an IOS router in SRST mode? (Choose three.) A. access-code B. dialplan-pattern C. ip source-address D. keepalive E. max-dn F. max-ephones Ans: C, E, F

Q17. Refer to Exhibit

You are debugging a problem on a SIP network and have run the debug ccsip messages command. One of the messages returned is shown in the exhibit. What information will the server return to the caller? A. the acceptable media type B. a list of acceptable media types C. a list of acceptable formats D. a correct directory number E. an acceptable language code Ans: C
<SIP_HEADERS>\ <ACCEPT_CONTACT value=\"*;+media=audio\"/>\ <ALLOW_EVENTS value=\"telephone-event\"/>\ <ACCEPT value=\"audio/basic\"/>\ <ACCEPT value=\"application/sdp\"/>\ <ACCEPT value=\"text/plain\"/>\ </SIP_HEADERS>\ <SDP_LINES>\

Q18. Which type of SIP responses would indicate that a server encountered an error in attempting to complete a SIP request?

A. 1 xx B. 3xx C. 4xx D. 5xx E. 6xx Ans: D 500 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

Q19.

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. proxy B. redirect C. registrar D. user agent client E. user agent server Ans: A

Q20. Which of the following three messages could be sent by the UAC in response to the 180 Ringing? (Choose three.) A. PR ACK B. ACK C. BYE D. CANCEL E. INVITE Ans: A, B, D (The PRACK request plays the same role as ACK but for provisional Responses. BYE and INVITE make no sense)

Q21. Which three attributes correctly describe aspects of MGCP? (Choose three.) A. peer-to-peer B. Master/Slave C. call preservation on gateway failover from one Cisco Unified Communications Manager server to another D. communication with Cisco Unified Communications Manager handled via a proxy server E. centralized dial plan management F. intelligent endpoints Ans: B, C, E

Q22. In 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. SIP B. H.323 C. MGCP D. SCCP Ans: C, D Q23. When implementing PRI backhaul for an MGCP gateway and Cisco Unified Communications Manager, the Q.921 data-link protocol is terminated on which device? A. Cisco Unified Communications Manager B. MGCP gateway C. signaling link terminal D. the IP end device, such as an IP phone Ans: B Q24. What occurs if the system clocks are not synchronized between the sender and receiver of an RTP stream? A. Packets can be placed in sequence but jitter cannot be compensated for. B. Packets cannot be reordered, because sequence and jitter cannot be compensated for. C. Jitter can be compensated for, but packets cannot be reordered if they arrive out of sequence. D. Packets may be reordered and jitter may be compensated for, because the timestamp is not related to the system time. E. When the RTP stream is opened, the sender and receiver synchronize their clocks before the stream commences so that packet sequencing and dejitter will function correctly. Ans: D Q25. On 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 seconds B. originating and terminating VoIP gateway for completing RSVP reservation setups in 10 seconds C. terminating VoIP gateway for completing RSVP reservation setups in 10 seconds D. VoIP gatekeeper for completing RSVP reservation setups in 10 seconds Ans: C Q26. Suppose the following command is configured in a gatekeeper: "bandwidth total default 64'. Which statement would be TRUE? A. This gatekeeper will admit up to 64 calls, regardless of the codec used. B. This gatekeeper will not admit any calls because all calls initially account of 128Kbps. C. This gatekeeper will admit a minimum of 4 calls using G.729 codec. D. This gatekeeper will admit up to 4 calls using G.729 codec. Ans: D

G.729=8kbit/sec x 2 (gatekeeper takes double bandwidth to compensate for in/out bound). Therefore 64/16=4 calls Q27. If enabled, the RSVP for LLQ feature will assign which two types of flows to the priority queue? (Choose two.) A. all RSVP bandwidth requests B. voice flows generated from Cisco IOS applications C. voice flows generated from third-party applications, such as Microsoft NetMeeting D. all traffic marked DSCP EF E. all traffic marked CoS 5 Ans: B, C The RSVP support for LLQ feature allows RSVP to classify voice flows and queue them into the priority queue (PQ) within the LLQ system while simultaneously providing reservations for nonvoice flows by getting a reserved queue. Q28. Which of these features are supported in RSVP Support for LLQ? (Choose three.) A. LLO Support on Tunnels B. Guaranteed Quality of Service C. Reserve resources for Low Latency and bandwidth guarantees D. LLQ on Frame Relay and ATM PVCs E. Controlled-Load Network Element Service Ans: B, C ,E
The RSVP Support for LLQ feature supports the following RFCs: RFC 2205, Resource Reservation Protocol RFC 2210, RSVP with IETF Integrated Services RFC 2211, Controlled-Load Network Element Service RFC 2212, Specification of Guaranteed Quality of Service RFC 2215, General Characterization Parameters for Integrated Service Network Elements

Q29. Users are complaining that the music on hold marketing files for this month are not being played when users are 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 restarted B. a new directory has been created for the new media files C. users have selected the correct MoH files for customer calls D. the new music files are in the correct format to be used with Cisco Unified Communications Manager E. the location of the new music files is what the MoH server expects Ans: A, D, E Users cannot specify the MoH file and a new directory is not required. Q30. Which 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 being placed on hold.

B. The audio stream and audio server used will be selected according to the configuration of the phone which is being used to place a caller on hold. C. The audio stream will be selected according to the configuration of the phone which is being used to place a caller on hold, and the audio server used will be selected according to the configuration of the phone being placed on hold. D. The audio stream will be selected according to the configuration of the phone being placed on hold and the audio server used will be selected according to the configuration of the phone which is being used to place a caller on hold. Ans: C Q31.Which two conditions will result in an H.323 gatekeeper receiving an ARQ from 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 keeplalive 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. Ans: B, E
ARQLocal zone messages that are sent by H.323 endpoints (usually gateways) to the Cisco gatekeeper. Gatekeepers receive ARQs from an endpoint if:

A local zone endpoint initiates a call. OR A local zone endpoint request permission to admit an incoming call.

Q32. Two H.323 gateways are engaged in an active call. How many RTP and RTCP packet streams exist between these two gateways? A. 2 B. 3 C. 4 D. 5 E. 6 Ans: C Video and audio are normally sent over different streams, we need to synchronize them at the receiver so that they play together. RTCP provides the information that is required for synchronizing the streams. So there are 2 streams per call. Q33. On a Cisco IOS MGCP gateway that is registered to Cisco Unified Call Manager, which MGCP message is initiated by the gateway? A. RQNT B. NTFY C. EPCF D. CRCX E. SETUP Ans: B

Call flow between two gateways example:

Q34. On a Cisco IOS MGCP gateway registered to CUCM, which of the FOUR messages are initiated by Call Manager?

A. Notification Request (RQNT) B. Notify (NTFY) C. Modify Connection (MDCX) D. Create Connection (CRCX) E. Audit Endpoint (AUEP) F. Reset In Progress (RSIP) Ans: A, C, D, E (NTFY and RSIP are initiated by gateways) Q35. On a Cisco IOS MGCP gateway registered to CUCM, which command can be issued by CUCM or gateway? A. Notification Request (RQNT) B. Create Connection (CRCX) C. Delete Connection (DLCX) D. Audit Endpoint (AUEP) E. Reset In Progress (RSIP) Ans: C

Refer to Exhibit for Q36 & Q37:

Concept:
huntstop channel is used on a shared line for incoming calls, rest of the channels are reserved for outgoing calls. Max-calls-per-button is used for maximum calls (incoming and outgoing) which can go through that line. Busy-trigger-per-button is used to busy out incoming calls after the parameter set has been reached. So if an ephone has busy-trigger-per-button set to 2 and the phone has two active calls then the third call will go to voicemail (if set), busy tone or forwarded to another number if call-forward no answer is set. Q36. What is the maximum number of inbound calls to 2001 before a Cisco Unified Communications Manager Express system returns a user busy tone to any additional calls? A. 3 B. 4 C. 5 D. 6 E. 7 Ans: 6 Octo-line channel hunting stops at channel 6: maximum 6 inbound calls for this octo-line ephone-dn Q37. What is the maximum number of inbound calls to ephone 1 before a Cisco Unified Communications Manager Express system returns a user busy tone to any additional calls? A. 3 B. 4 C. 5 D. 6 E. 7 Ans: B (busy-trigger-per-button is for incoming calls)

Refer to Exhibit for Q38 & Q39: ! ephone-dn 1 octo-line number 2001 ! ephone 1 max-calls-per-button 5 busy-trigger-per-button 4 button 1:1 ! ephone 2 Max-calls-per-button 6 busy-trigger-per-button 3 button 1:1 !
Q38. What is the maximum or total number of calls to ephone 2 before a Cisco Unified Communications Manager Express system returns a user busy tone to any additional calls? A. 3 B. 4 C. 5 D. 6 E. 7 Ans: B (Max-calls-per-button is used for maximum calls (incoming and outgoing) which can go through that line)

Q39. What is the maximum number of inbound calls to 2001 before a Cisco Unified Communications Manager Express system returns a user busy tone to any additional calls? A. 3 B. 4 C. 5 D. 6 E. 7 F. 8 Ans: E (ephone 1 + ephone 2 busy-trigger-per-button value is 4+3=7. Remember there is no huntstop channel command in ephone-dn 1) Q40. Which two analog voice interfaces support ground-start? (Choose two.) A. FXS B. E&MType l C. E&MType ll D. E&MType lV

E. FXO Ans: A, E (Cisco does not support E&M Type IV)

Q41. Which signaling method cannot solve the FXO disconnect problem? A. power denial B. tone-based supervisory disconnect C. pulse dial D. ground-start signaling E. battery reversal Answer: C Q42. Which three signaling types are not used by analog E&M circuits as start dial supervision protocols? (Choose three) A. delay dial B. wink-start C. ground-start D. wink-start Feature Group D E. immediate-start F. pulse dial Ans: C, D, F (On E&M circuits, 3 main Start Dial Supervision protocols are: delay dial, wink-start, and immediate-start)
The signaling method selection for type depends on the connection that you are making. The E&M interface allows connection for PBX trunk lines (tie lines) and telephone equipment. The options are as follows: e&m-immediate-start specifies no specific offhook and onhook signaling. e&m-delay specifies that the originating endpoint sends an offhook signal and then and waits for an offhook signal followed by an onhook signal from the destination. e&m-fgd specifies E&M Type II Feature Group D. e&m-wink-start specifies that the originating endpoint sends an offhook signal and waits for a wink signal from the destination. fgd-eana specifies Group D Exchange Access North American (EANA) signaling.

Q43. Which three are valid T1 CAS types? (Choose three.) A. E&M signaling B. semicompelled signaling C. loop-start signaling D. line signaling E. Group 1 signaling F. ground-start signaling Ans: A, C, F (Channel Associated Signaling (CAS) is also referred to as Robbed Bit Signaling)

Refer:

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

Q44. Which R2 signaling element passes address information such as calling and called party numbers? A. pulse signaling B. delay dial signaling C. line signaling D. interregister signaling E. out-of-band signaling

Ans: D
(The concept of address signaling in R2 is slightly different from that used in the other CAS systems previously discussed. In the case of R2, the exchanges are considered registers, and the signaling between these exchanges is called inter-register signaling. Inter-register signaling uses forward and backward in-band MF signals to transfer called and calling party numbers as well as the calling party category)

Q45. How many frames are contained in one multiframe within an SF format? A. 4 B. 8 C. 15 D. 16 E. 30 F. 32 Ans: D Q46. How many channels on a voice E1 circuit carry PCM-encoded voice traffic? A. 16 B. 28 C. 29 D. 30 E. 31 Ans: D Q47. According to the IEEE 802.3af PoE standard, what is the maximum power (in watts) that is delivered to a power consuming device? A. 6.3 B. 14.5 C. 15.4 D. 20 E. 22.5 F. 25.4 Ans: C

(The original IEEE 802.3af-2003 PoE standard provides up to 15.4 W of DC power (minimum 44 V DC and 350 mA) to each device. Only 12.95 W is assured to be available at the powered device as some power is dissipated in the cable). Q48. What is the complete name of LLDP-MED, an enhancement to the vendor-neutral LLDP that is supported on Cisco switches? A. Link Layer Discovery Protocol-Media Endpoint Discovery B. Link Layer Discovery Protocol-Media Enhancement Discovery C. Link Layer Discovery Protocol-Media Enhancement Delivery D. Link Layer Discovery Protocol-Multiple Enhancement Delivery E. Link Layer Discovery Protocol-Multiple Endpoint Discovery Ans: A (LLDP-MED and Cisco Discovery Protocol are closely related protocols. They are similar in many ways, including operation and the information they carry. The most important benefit with LLDP-MED is that it is an open standard) Q49. Which of these is not a valid switchback method for SCCP hardware conference bridges? A. immediate B. never C. graceful D. guard E. uptime Answer: B switchback method {graceful | guard [guard-timeout] | immediate |uptime uptime-timeout}

Q50. Which two are valid switchover methods for SCCP hardware conference bridges? A. immediate B. guard C. uptime delay D. Schedule time E. graceful F. never Ans: A, E GracefulThe Cisco Unified Call Manager switchover happens only after all the active sessions are terminated gracefully. This is the default method. ImmediateRegardless of whether there is an active connection or not, the SCCP client switches over to one of the secondary Cisco Unified Call Managers immediately. If the SCCP Client is not able to connect to a secondary Cisco Unified Call Manager, it continues to poll for a Cisco Unified Call Manager connection.

Q51. Which is the default switchback method for an SCCP hardware transcoder when a higherpriority CUCM becomes available again? A. immediate B. never C. graceful D. uptime E. guard Ans: E (Default method is guard, with a timeout value of 7200 seconds. Graceful is optional) Q52. Which telephony signaling type cannot be configured for a Cisco IOS MGCP gateway on Cisco Unified Communications Manager? A. T1 PRI B. analog FXO C. analog E&M D. T1 CAS E&M delay dial E ISDN BRI Ans: C
The following types of interfaces on the gateway are supported: FXS analog interfacesFor connecting to the PSTN or analog phones FXO analog interfacesFor connecting to the PSTN or PBXs T1 CAS digital interfacesFor connecting to the PSTN or PBXs T1 PRI digital interfacesFor connecting to PBXs and central offices (COs)

Q53. Which two codecs provide built-in VAD? (Choose two) A. G.711 U law B. G.722-64k C. G.723.1 ANNEX-A D. G.726 E. G.729 F. G.729 ANNEX-B Ans: C, F Q54. Which statement about the G.729 codec is correct? A. G.729 and G.729A are both high-complexity codec 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 Ans: D

Q55. Which codecs are considered to have medium complexity on Cisco IOS voice gateways? (Choose three) A. G.711 a-law B. G.711 mu-law C. G.723 D. G.728 E. G.729 F. G.729 ANNEX-A ANNEX-B

Ans: A, B, F

Q56. Which codecs are considered to have high complexity on Cisco IOS voice gateways? (Choose three)

A. G.711 a-law B. G.711 mu-law C. G.723 D. G.728 E. G.729 F. G.729 ANNEX-B Ans: C, D, F

Medium Complexity (4 calls / dsp) G.711 (a-law and m -law) G.726 (all versions) G.729a, G.729ab (G.729a AnnexB) Fax-relay

High Complexity ( 2 calls / dsp) G.728 G.723 (all versions) G.729, G.729b (G.729-AnnexB) Fax-relay

Q57. Which statement about the G.729 codec 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. Ans: E

Q58. On a Cisco IOS MGCP PRI gateway, what is the maximum configurable length of time for a scheduled switchback to a higher-priority Cisco Unified Communications Manager? A. 6 hours B. 12 hours C. 18 hours D. 24 hours E. 48 hours Ans: D ccm-manager switchback {graceful | immediate | schedule-time hh:mm | uptime-delay minutes} <1-1440> Delay time (minutes) 1440/60=24 hours

Q59. Which of these is an invalid switchback method for a Cisco IOS MGCP PRI gateway in case a higher-priority CUCM returns to active service? A. guard B. schedule-time C. graceful D. uptime delay E. immediate Ans: A ccm-manager switchback {graceful | immediate | schedule-time hh:mm | uptime-delay minutes}

Q60. Which of these is a default switchback method for a Cisco IOS MGCP PRI gateway in case a higher-priority CUCM returns to active service? A. graceful B. schedule-time C. guard D. uptime delay E. immediate Ans: A Q61. Which 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. Ans: E

To configure Unicast MOH for HQ site and Multicast MOH for other sites if you only have one CM server (Publisher) and therefore only one MOH Server. under "Media Resources" in CUCM server make sure you check 3 boxes. (1-Allow Multicasting, 2-Enable Multicast Audio Sources on this MOH Server, 3-Use Multicast for MOH Audio)

Q62. Which 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.

Ans: C Q63. CUCM Server A and CUCM Server B are in same cluster. The cluster has a total of four registered conference bridges: two software conference bridges (one from each server) and two Hardware conference bridges. All four bridges are registered to Server B as the Primary callprocessing node and Server A as backup. If Administrator accidentally deactivated the Cisco IP Voice Media Streaming Application service on Server B, what will happen to conference 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 Bridge will deregister, and both Hardware bridges will remain registered to Server B. E. Both Software Bridge will deregister, and both Hardware bridges will registered to Server A Ans: C Q64. CUCM Server A and CUCM Server B are in same cluster. The cluster has a total of four registered conference bridges: two software conference bridges (one from each server) and two Hardware conference bridges. All four bridges are registered to Server B as the Primary callprocessing node and Server A as backup. If Administrator accidentally deactivated the Cisco Call Manager service on Server B, what will happen to conference 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 Bridge will deregister, and both Hardware bridges will remain registered to Server B. E. Both Software Bridge will deregister, and both Hardware bridges will registered to Server A Ans: A

Q65. Which string is not valid route pattern on CUCM? A. 123@ B. 123. C. 123* D. 123$ E. 123? Ans: D Q66. Which CUCM route pattern character represents Zero or more occurrences of the previous digit or wildcard? A.! B.+ C.* D. . E.? Ans: E Q67. Which CUCM route pattern character represents Zero or more occurrences of the digits in the range zero to nine? A.! B.+ C.* D. . E.? Ans: A Q68. Which CUCM route pattern character represents one or more occurrences of the previous digit or wildcard? A.! B.+ C.* D. . E.? Ans: B

Q69. Refer to Exhibit.

! Voice-port 0/2/0 Description FXO port Connection plar opx 1001 Caller-id enable !
A. connection plar opx is wrong it should be replaced with connection plar. B. 1001 is configured for delay 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. Ans: C
The two ring delay is for the caller ID to be received. This is a very common standards industry practice. Without two rings first, the customer may not see the caller id right away.

Q70. What is the default method of handling an H.323 connection from unknown devices on CUCM? A. Cisco Unified Call Manager accepts incoming H.323 connections from unknown devices B. Cisco Unified Call Manager ignores incoming H.323 connections from unknown devices C. Cisco Unified Call Manager rejects incoming H.323 connections from unknown devices by sending an H.323 reject D. Cisco Unified Call Manager rejects incoming H.323 connections from unknown devices by sending an H.323 disconnect E. Cisco Unified Call Manager rejects incoming H.323 connections from unknown devices by closing the TCP socket. Ans: E Q71. What is the default method of handling an H.323 connection from unknown devices on a Cisco IOS H.323 gateway? A. Cisco IOS H.323 gateway accepts incoming H.323 connections from unknown devices B. Cisco IOS H.323 gateway ignores incoming H.323 connections from unknown devices C. Cisco IOS H.323 gateway rejects incoming H.323 connections from unknown devices by sending an H.225 reject D. Cisco IOS H.323 gateway rejects incoming H.323 connections from unknown devices by sending an H.225 disconnect E. Cisco IOS H.323 gateway rejects incoming H.323 connections from unknown devices by closing the TCP socket. Ans: A Q72. An H.225 call setup arrives at Cisco Unified Communications Manager for a directory number on an IP phone that is engaged in an active conversation. If call waiting is disabled for this directory number and none of the Call Forward settings are defined, which H.225 disconnect reason code will be sent to the originating H.323 gateway?

A. No Route To Destination B. Normal Call Clearing C. Subscriber Absent D. User Busy E. Network Busy Ans: D Q73. When an H.225 call setup arrives at Cisco Unified Communications Manager for a directory number on an IP phone with a partition that is not reachable by the H.323 gateway calling search space, which H.225 disconnect reason code will be sent to the originating H.323 gateway? A. No Route To Destination B. Unallocated (Unassigned) Number C. Number unreachable D. Number available but out of reach E. Network Busy Ans: B Q74. When a call arrives from the PSTN on a Cisco IOS MGCP PRI gateway that is registered to CUCM, destined to an IP phone directory number with a partition that is not reachable by the MGCP gateway calling search space, which event will take place? A. Cisco Unified Call Manager will send the call to Call Forward Busy destination that is configured on the IP Phone. B. Cisco Unified Call Manager will disconnect the call with an MGCP DLCX message C. Cisco Unified Call Manager will disconnect the call with a Q.931 cause of No Route To Destination D. Cisco Unified Call Manager will disconnect the call with a Q.931 cause of Unallocated (Unassigned) Number E. Cisco Unified Call Manager will disconnect the call with a Q.931 cause code of 420, which means Bad Extension Ans: D Q75. When a call arrives from the PSTN on a Cisco IOS MGCP PRI gateway that is registered to CUCM, destined for a directory number on an IP phone that is engaged in an active conversation. If call waiting is disabled for this directory number and none of the Call Forward settings are defined, which event will take place? A. Cisco Unified Call Manager will disconnect the call with an MGCP RSIP message. B. Cisco Unified Call Manager will disconnect the call with an MGCP DLCX message with a cause of Busy C. Cisco Unified Call Manager will disconnect the call with a Q.931 cause of User Busy D. Cisco Unified Call Manager will disconnect the call with a Q.931 cause of Temporary Failure E. Cisco Unified Call Manager will disconnect the call with a Q.931 cause code of 486, which means Busy Here Ans: C

Q76. How many bits in an 802.1Q tagged Ethernet frame are used for 802.1p priority? A. 3 B. 4 C. 5 D. 6 Ans: A 16 bits 3 bits 1 bit 12 bits

TCI TPID PCP CFI VID

Q77.

A. A B. B C. C D. D Ans: A

Q78.

A. A B. B C. C D. D Ans: D Q79.

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

Q80.

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

Q81.

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

Q82. Your client has a business requirement that mandates exact DTMF durations being passed end to-end across an H.323 VoIP infrastructure. Which two DTMF relay methods meet the client requirement? (Choose two.) A. Cisco RTP B. H.245 signal C. H.245 alphanumeric D. RTP-NTE E. H.225 Notify F. in-band voice Ans: B, D
The H.245 signaling channel is a reliable channel, so the packets that transport the DTMF tones are guaranteed to be delivered. Until now, the NTE method of DTMF relay was available only for Session Initiation Protocol (SIP) and Media Gateway Control Protocol (MGCP) gateways. The H.323 Dual Tone Multifrequency Relay Using Named Telephone Events feature adds support for this method for H.323 gateways.

Q83. CUCM generates different types of alarms to indicate system or process related problems. Code Yellow is one of these alarms. Which of these system process exceptions will trigger a code Yellow alarm on CUCM. A. When hard drive fails B. When there is a memory leak C. When CUCM application generates a core dump D. When a database replication problem arises E. When calls are throttled because of an unacceptably high delay in call handling Ans: E
Call throttling allows Cisco Unified CallManager to automatically throttle (deny) new call attempts when it determines 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 to dial tone (the interval users experience from going off hook until they receive dial tone). Cisco Unified CallManager uses the values that are specified in the call-throttling-related parameters to evaluate the possibility of a delay to dial tone and also to determine when conditions no longer necessitate call throttling. When throttling is necessary to prevent excessive delay to dial tone, Cisco Unified CallManager enters a Code Yellow state, and new call attempts are throttled (denied).

Q84. In which two circumstances would CUCM accept inbound H.323 calls from unknown IP hosts? (Choose two) A. When inbound H.323 calls are routed via gatekeeper-controlled trunks. B. When inbound calls are routed via intercluster trunks C. After administrators have changed the CUCM clusterwide service parameter of Accept unknown TCP connection to true. D. When inbound H.323 calls are routed via non gatekeeper-controlled trunks. E. When inbound calls are routed using H.323 fast start F. After administrators have changed the CUCM clusterwide service parameter of Accept unknown Caller ID Flag to true. Ans: A, C

Q85. The Cisco UMR feature allows Cisco Unity to take outside caller messages while their Exchange Server is unavailable. Which two statements about Cisco UMR are correct? (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 handle 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 Cisco Unity installation. F. During an Exchange outage, messages to the unaddressed message distribution lists appear in Cisco UMR and can be accessed by all members of the list. Ans: E, F Specifically, the space available in the C:\Commserver\UnityMTA directory. This directory is controlled by a registry setting. During 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 Q86. Cisco Unity extends a number of schema object classes in Microsoft Active Directory during the schema extension process. Which three object classes are extended by the Cisco Unity schema extension process? (Choose three.) A. user B. computer C. domain D. organizational unit E. group F. contact Ans: A, E, F

Q87. Refer to the Exhibit Using information that is provided in the Cisco IOS gatekeeper configuration and the show gatekeeper endpoint output, 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 gwtype-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. Ans: C

show gatekeeper gw-type-prefix - Used to verify E.164 prefix registrations on the gatekeeper show gatekeeper zone prefix | status - Used to verify zone status and configuration parameters show gatekeeper endpoints Used to verify the E.164 and H.323 alias registered with the gatekeeper
Router#show gatekeeper gw-type-prefix GATEWAY TYPE PREFIX TABLE =========================== Prefix:12#* (Default gateway-technology) Zone sj-gk master gateway list: 172.21.13.11:1720 sj-gw1 172.21.13.22:1720 sj-gw2 (out-of-resources) 172.21.13.33:1720 sj-gw3 Zone sj-gk prefix 408....... priority gateway list(s): Priority 10: 172.21.13.11:1720 sj-gw1 Priority 5: 172.21.13.22:1720 sj-gw2 (out-of-resources) 172.21.13.33:1720 sj-gw3

Q88. Refer to Exhibit.

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. Ans: B (This is the correct configuration for T1 PRI)

Q89. Which set of SIP headers is mandatory in SIP requests? A. Call-ID, Contact, User-Agent, RSeq, SDP B. Allow, Supported, Via, From, To, CSeq C. Via, From, To, Call-ID, CSeq, Contact D. Content-Type, Content-Length, Session-Expires, Via, From E. Req URI, From, Via, To, CSeq Ans: C {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}
69.7.163.154:5060;branch=z9hG4bK400fc6e6 From: "8069664170" <sip:8069664170@69.7.163.154>;tag=as42e2ecf6 To: <sip:01150259917040@67.135.76.4> Contact: <sip:8069664170@69.7.163.154> Call-ID: 2485823e63b290b47c042f20764d990a@69.7.163.154 CSeq: 102 INVITE User-Agent: MatrixSwitch Date: Thu, 2INVITE sip:01150259917040@67.135.76.4 SIP/2.0 Via: SIP/2.0/UDP 2 Dec 2005 18:38:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 268 Via contains the address at which the originator is expecting to receive responses to this request .Mandatory

To contains a display name and a SIP URI towards which the request was originally directed. Mandatory Display names are described in RFC 2822 From also contains a display name and a SIP URI that indicate the originator of the request. The From also contains a tag parameter which is used for identification purposes. Mandatory Call-ID contains a globally unique identifier for this call. Mandatory CSeq or Command Sequence contains an integer and a method name. The CSeq number is incremented for each new request within a dialog and is a traditional sequence number. Mandatory Contact contains a SIP URI that represents a direct route to the originator usually composed of a username at a fully qualified domain name (FQDN). While an FQDN is preferred, many end systems do not have registered domain names, so IP addresses are permitted. The Contact header field tells other elements where to send future requests. Max-Forwards serves to limit the number of hops a request can make on the way to its destination. It consists of an integer that is decremented by one at each hop. Content-Type contains a description of the message body. Mandatory Content-Length contains an octet (byte) count of the message body.

Q90. Which 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. Ans: C Q91. Which 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.

Ans: E
Q92. Which two mechanisms can be used to detect SIP calls that are hung or stuck in an incomplete state? (Choose two.) A. SDP time stamps and version number B. PRACK (RSeq) C. session timer D. periodic hold/resume E. OOD Refer F. RTP and RTCP inactivity monitoring Ans: C, F

We can RTP / RTCP Media Inactive Call Detection feature in the GW to clear hung calls. The SIP Session Timer Support feature adds the capability to periodically refresh Session Initiation Protocol (SIP) sessions by sending repeated INVITE requests. The repeated INVITE requests, or reINVITEs, are sent during an active call leg to allow user agents (UA) or proxies to determine the status of a SIP session. Without this keepalive mechanism, proxies that remember incoming and outgoing requests (stateful proxies) may continue to retain call state needlessly. If a UA fails to send a BYE

message at the end of a session or if the BYE message is lost because of network problems, a stateful proxy does not know that the session has ended. The re-INVITES ensure that active sessions stay active and completed sessions are terminated.
Q93. Which statement correctly describes symmetric signalling 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. Ans: E Q94. Refer to Exhibit.

The line that is shown in the exhibit appeared in the Cisco Unified Communications Manager trace for a SIP IP phone 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.

Ans: B
Q95. Which statement about the "Unknown Caller ID" service parameter in Cisco Unified Communications Manager Configuration is true? A. This parameter defines a numeric string to be displayed to the called party on inbound calls that arrived with no caller ID information. B. This parameter designates a numeric string to be displayed to the called party for outbound calls without caller ID information. C. This parameter defines a numeric or a text string to be displayed to the called party for inbound calls that arrived with no caller ID information. D. This parameter designates a numeric or a text string to be displayed to the called party for outbound calls without caller ID information. E. This parameter defines a numeric string to be displayed to the called party on inbound and outbound calls with no caller ID information. Ans: A Q96. Which two media resources are not required by Cisco Unified Communications Manager for outbound earlyoffer support on SIP trunks? (Choose two.) A. annunciator B. software-based MTP in Cisco IOS gateways C. hardware-based MTPs in Cisco IOS gateways D. software-based MTPs using the Cisco IP Voice Media Streaming Application on Cisco MCS E. Cisco Unified Border Element with H.323-to-SIP interworking enabled Ans: A, E.

In an Early Offer, the session initiator (calling device) sends its capabilities (for example, codecs supported) in the SDP contained in the initial Invite (thus allowing the called device to choose its preferred codec for the session). In Early Media, the media channel is setup before the call is established. This can be used to reduce the effects of audio cut-through delay or to provide a network-based voice message to the caller. In most sip calls, Media is established after the calling party receives a 200 OK. With Early media, Media is established in 183 with SDP, much before 200 OK is sent. This allows providers to play announcement or MOH or whatever they choose to play before sending 200 OK.

Q97. There are several methods to transport DTMF digits between SIP endpoints. Which three methods are supported by Cisco Unified Communications Manager? (Choose three.) A. Unsolicited Notify B. INFO C. KPML D. RFC 2833 in-band signal tone/event E. Cisco RTP-NTE F. H.245 alphanumeric Ans: ACD (RTP-NTE and H.245 are H.323 counterparts)
The SIP phone initiates a payload type response when the user enters a number on the keypad. The SIP phone transfers the DTMF in-band digit (per RFC 2833) to the MTP device. KPML (key press Markup Language) Session Initiation Protocol (SIP)". The event package uses SUBSCRIBE messages and allows for XML documents that define and describe filter specifications for capturing key presses (DTMF Tones) entered at a presentation-free User Interface SIP User Agent (UA). The event package uses NOTIFY messages and allows for XML documents to report the captured key presses (DTMF tones).

Q98. According to the Cisco QoS SRND guide, cRTP is recommended on which link speed? A. lower than or equal to 10 Mb/s B. lower than or equal to 384 kb/s C. lower than or equal to 1.544 Mb/s D. lower than or equal to 768 kb/s E. lower than or equal to 512 kb/s Ans: D

Q99. Which Cisco IOS CLI command can be used to identify the high jitter level of an RTP stream on a Cisco IOS voice gateway? A. show call active voice brief B. show voip rtp connections C. show voice dsp detailed D. show voice call summary E. show policy-map interface Ans: A

Q100. According to RFC 3551, where default mappings between RTP payload type numbers and encodings are defined, which RTP payload type corresponds to G.711 a-law encoding? A. 0 B. 1 C. 8 D. 13 E. 18 Ans: C
A-law encoding takes a 13-bit signed linear audio sample as input and converts it to an 8 bit value. -law encoding takes a 14-bit signed linear audio sample as input, increases the magnitude by 32 (binary 10000), and converts it to an 8 bit value.

Q101. According to RFC 3551, where default mappings between RTP payload type numbers and encodings are defined, which RTP payload type corresponds to encoded packets that are triggered by silence on a call with voice activity detection? A. 0 B. 13 C. 15 D. 18 Ans: B 13 CN audio 1 8000 Comfort noise RFC 3389,

Q102. Which version of MGCP is used on a Cisco IOS MGCP gateway that is registered to Cisco Unified Communications Manager? A. 0.0 B. 0.1 C. 1.0 D. 1.1 E. It varies between Cisco Unified Communications Manager versions. Ans: B
Example: Router# mgcp call-agent 10.0.0.21 mgcp 0.1 (MGCP call-agent: none Initial protocol service is MGCP 0.1)

Q103. On a Cisco IOS MGCP gateway, which DTMF relay method uses MGCP NTFY messages to send digits to Cisco Unified Communications Manager? A. Cisco B. NSE C. NTE-CA D. NTE-GW E. out-of-band Ans: E

Explanation: 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 voice codecs. low-bitrate Configures DTMF relay to be used with only low-bit-rate voice codecs, such as G.729. cisco Real-time Transport Protocol (RTP) digit events are encoded using a proprietary format similar to frame relay as described in the FRF.11 specification. The events are transmitted in the same RTP stream as nondigit 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 voice samples, using the payload type that is configured using the mgcp tse payload command. out-ofband Media gateway control protocol (MGCP) digit events are sent using NTFY messages to the call agent (CA), which plays them on the remote GW using RQNT 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 voice samples. The payload type is negotiated by the GWs before use. The configured value for payload type is presented as the preferred choice at the beginning of the negotiation. nte-ca Identical to the nte-gw keyword behavior except that the CA's local connection options a: line is used to enable or disable DTMF relay. Q104. Which SCCP message is used to instruct to an SCCP IP phone the remote IP address and port number to send RTP packets? A. Station IP Port message B. Station Open Receive Channel message C. Station Start Media Transmission message D. Station Call Information message E. Station Open Logical Channel message Ans: C

Q105. An IP phone user just answered an incoming call by lifting the handset. Assuming that the IP phone uses SCCP, which SCCP message will Cisco Unified Communications Manager transmit to this called IP phone immediately after receiving notification about the off-hook event? A. Station Media Port List message B. Station Set Ringer message C. Station Stop Tone message D. Station Start Media Transmission message E. Station Open Receive Channel message Ans: B

Q106. Which SCCP message is used by an IP phone to inform Cisco Unified Communications Manager about the IP address and port number to be used for an incoming RTP stream? A. Station Capability Response message B. Station IP Port message C. Station Open Receive Channel ACK message D. Station Media Reception ACK message E. Station Start Media Transmission message Ans: C

Q107. When an IP phone that is using SCCP places an active call on hold, which SCCP message will be transmitted from the phone to Cisco Unified Communications Manager? A. Station On Hold message B. Station Keypad Button message C. Station Close Receive Channel message D. Station Stop Media Transmission message E. Station Softkey Event message Ans: E Q108. What is the proper Cisco IOS CLI command to configure an analog FXS port to be controlled by Cisco Unified Communications Manager using SCCP? A. dial-peer voice 1 pots port 1/0 service skinny B. dial-peer voice 1 pots port 1/0 service sccp C. dial-peer voice 1 pots port 1/0 service stcapp D. dial-peer voice 1 pots port 1/0 service sccpapp E. dial-peer voice 1 pots port 1/0 application sccp Ans: C

Q109. Refer to Exhibit

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

A. II, I, III, V, IV B. III, I, II, V, IV C. III, II, I, V, IV D. II, III, I, V, IV E. V, III, II, I, IV F. V, II, III, I, IV Ans: C Explanation: 1. Called number (DNIS) with the incoming called-number command 2. Calling Number (ANI) with the answer-address command 3. Calling Number (ANI) with the destination-pattern command 4. 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. Q110. Which two attributes are not used by a Cisco IOS router in the inbound dial-peer selection process? (Choose two.) A. default dial-peer 0 B. called number with the destination-pattern command of each dial peer C. calling number with the destination-pattern command of each dial peer D. calling number with the answer-address command of each dial peer E. called number with the incoming called-number command of each dial peer F. called number with the answer-address command of each dial peer G. voice port that is associated with an incoming call Ans: B, F

Q111. Refer to the Exhibit.

When an inbound call with a calling number of 1001 and a called number of 2112 arrives at a Cisco IOS router with these dial peers, what is the correct order of dial-peer matching, from highest to lowest priority? A. II, III, I, IV B. II, IV, III, I C. III, IV, II, I D. III, II, IV, I E. II, III, IV, I Ans: D Q112. Refer to Exhibit

When an outbound call with a calling number of 2112 and a called number of 1001 is placed through a Cisco IOS router with these dial peers, what is the correct order of dial-peer matching, from highest to lowest priority? A. I, II, III, IV B. II, I, III, IV C. II, I, IV, III D. I, II, IV, III E. II, IV, I, III Ans: B
The method a router uses to select an outbound dial peer depends on whether ISDN DID is configured in the inbound POTS dial peer. If DID is not configured in the inbound POTS dial peer, the router collects the incoming dialed string digit by digit. As soon as one dial peer is matched, the router immediately places the call using the configured attributes in the matching dial peer

Q113. What is the default DTMF relay for Cisco Unity Express when integrated via SIP? A. RTP-NTE B. SIP Notify C. SIP INFO D. in-band audio E. SIP Subscribe/Notify Ans: B
dial-peer voice 6800 voip destination-pattern 5... session protocol sipv2 session target ipv4:a.3.6.127 dtmf-relay sip-notify codec g711ulaw no vad In the SIP-NOTIFY DTMF relay mechanism, the Cisco Unity Express application subscribes internally to Cisco CME call control for telephony events (in this case, the event is the pressing of a digit) as soon as the call is established. Cisco CME accepts this subscription and sends a NOTIFY to Cisco Unity Express whenever a digit is pressed on the phone.

Q114. Refer to Exhibit.

This Cisco Unified Communications Manager trace shows a SIP message that is sent by a SIP Cisco Unified IP Phone 7965 to Cisco Unified Communications Manager. Which of these regarding the content of this SIP message is correct? A. phone registration message to the primary Cisco Unified Communications Manager B. keepalive message to the primary Cisco Unified Communications Manager C. phone registration message to the secondary Cisco Unified Communications Manager during a server failover D. keepalive message to the secondary Cisco Unified Communications Manager E. phone registration message to the primary Cisco Unified Communications Manager during fallback Ans: D

Q115. Refer to the Exhibit.

This Cisco Unified Communications Manager trace shows a SIP message that was sent by a SIP Cisco Unified IP Phone 7965 to Cisco Unified Communications Manager. Which of these about the content of this SIP message is correct? A. phone registration message to the primary Cisco Unified Communications Manager B. keepalive message to the primary Cisco Unified Communications Manager C. phone registration message to the secondary Cisco Unified Communications Manager during a server failover D. keepalive message to the secondary Cisco Unified Communications Manager E. phone registration message to the primary Cisco Unified Communications Manager during fallback Ans: B

Q116. A customer purchased 10,000 phone license units for a Cisco Unified Communications Manager cluster. How many phone license unit overdrafts are permitted in this Cisco Unified Communications Manager cluster? A. 200 B. 500 C. 700 D. 1000 E. Phone license unit overdrafts are never permitted. Ans: B 5% overdraft. 5% of 10.000 = 500

An overdraft condition occurs when more licenses are used than available but the amount of exceeding licenses is in an acceptable range. (5 percent overdraft is permitted.)

Q117. Refer to Exhibit.

The error alert that is shown in the exhibit is seen in the "Event Viewer--Application Log" on Cisco Unified Presence. 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. Ans: A
This alert occurs when the percentage of used disk space in the log partition exceeds the configured high water mark. When this alert gets generated, LPM deletes files in the log partition (down to low water mark) to avoid running out of disk space.

Q118. Refer to Exhibit.

The log was captured for a Cisco Unified Presence client that is not able to perform desk phone control to a Cisco 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. Ans: B, D Q119. Which network port is not used by the Cisco Unified Presence client? A. TCP port 143 B. TCP port 2000 C. TCP port 2748 D. UDP port 69 E. TCP port 443 F. TCP port 389 Ans: B

143 - used by IMAP 2748 - used by CTI Gateway 69 - used by TFTP 443 - used by HTTPS 389 - used by LDAP

Q120. When using the Local Route Group feature in Cisco Unified Communications Manager, in which two levels can you apply the called party transformation pattern? (Choose two.) A. device pool B. gateway C. route pattern D. route group E. route list F. service parameter Ans: A, B (verified in CUCM) Q121. Refer to Exhibit.

Which dial peer will the Cisco IOS voice gateway match if an incoming call with a calling number of 100 and called number of 101 arrives at this T1 PRI port? A. dial-peer voice 2 B. dial-peer voice 3 C. dial-peer voice 5 D. dial-peer voice 8 E. dial-peer voice 0 Ans: B

Q122. Refer to Exhibit.

Which dial peer will the Cisco IOS voice gateway match if an incoming call with a calling number of 100 and called number of 101 arrives at this T1 PRI port? A. dial-peer voice 2 B. dial-peer voice 3 C. dial-peer voice 5 D. dial-peer voice 8 E. dial-peer voice 0 Ans: D

Q123. Which Cisco IOS command and configuration mode can be used to force a Cisco IOS voice gateway to use TCP as the transport protocol for SIP?

A. router(config)#sip transport tcp B. router(conf-voi-serv)#no sip transport udp C. router(conf-serv-sip)#no transport udp D. router(conf-serv-sip)#transport tcp E. router(config-sip-ua)#no transport udp Ans: E

Q124. Refer to Exhibit.

Which two statements about this debug message that was captured from a Cisco IOS H.323 gateway are correct? (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 . Ans: A, C

Q125. Refer to Exhibit

The exhibit shows how MOH Server A and MOH Server B are associated with Phone A and Phone B. If Phone A presses the Hold softkey during an active call with Phone B using the G.711 mu-law codec, which two statements 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. Ans: B, E Q126. When calls are placed by certain Cisco Unified Communications Manager supplementary services, the Local Route Group feature will be bypassed. Which of these does not belong to the supplementary services? A. Call Back B. Call Forward C. Message Waiting Indicator D. Mobility Follow Me E. Path Replacement Ans: B

Many supplementary services can originate calls. When this happens, the local route group gets skipped. The following features can initiate calls: CallBack MWI Mobility (FollowMe) Path Replacement

Q127. Phone 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 for each device pool. Phone B has set CFA to Phone C; Phone C has set CFNA to a PSTN number. When Phone A 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. Ans: A Q128. Refer to Exhibit.

The exhibit shows the T.30 message exchanges that resulted in a single page fax call failure. Which T.30 message sequence will result in a successful fax transmission? A. PPS, EOP, PPR, PPS, EOP, NSF, DCN B. MPS, EOP, PPR, PPS, EOP, MCF, DCN C. PPS, EOP, RTP, PPS, EOP, NSF, DCN D. PPS, EOP, PPR, PPS, EOP, MCF, DCN E. PPS, EOP, NSF, DCN Ans: D
Partial page signal (PPS) message is used in ECM to indicate the end of a partial block. Partial page request (PPR) if the partial page is not received correctly. EOP (end of procedure)- Indicates the end of a complete page of fax information and signals that no further pages are to be sent. MCF (message confirmation) -Indicates that a message has been satisfactorily received. DCN (disconnect)-Ends the fax call and requires no response.

Q129. Refer to Exhibit.

The debug outputs that are shown in the exhibit were collected at the terminating Cisco IOS gateway for a fax call 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. Ans: A, C Q130. Which MGCP message is used to indicate fax switchover in a call agent-controlled T.38 fax relay? A. CRCX B. NSE C. NTE D. NTFY E. MDCX Ans: E
EPCF EndpointConfigurationspecifies the encoding of the signals that will be received by the endpoint. RQNT NotificationRequestrequests the gateway to send notifications upon the occurrence of specified events in an endpoint. NTFY Notifysent by the gateway in compliance with RQNT when a triggering event occurs. CRCX CreateConnectioncreates a connection between two endpoints. MDCX ModifyConnectionmodifies the characteristics of a gateway's "view" of a connection. This "view" of the 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 collects

statistics on the execution of the connection. DeleteConnection (from the gateway)issued by the media gateway to clear a connection, for example because it has lost the resource associated with the connection, or because it has detected 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 that relate to a Call for an endpoint or terminate in a given endpoint. AUEP AuditEndpointused by the call agent to find out the status of a given endpoint. AUCX AuditConnectionused by the Call Agent to retrieve the parameters attached to a connection. RSIP RestartInProgressused by the gateway to signal that an endpoint, or a group of endpoints, is put in-service or out-of-service.

Q131. Refer to Exhibit.

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 further processing. The IP phone operator has reported that the calling number and name are absent for all incoming PSTN calls. Which configuration will resolve this issue? A.

B. 65

C.

D.

Ans: D Q132. Refer to Exhibit.

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

A. in-band voice B. RTP-NTE C. SIP KPML D. SIP Notify E. Cisco RTP F. RTP-NSE Ans: B, D

RTP Use Payload Type NTE with the default value of 101.

Q133. Refer to Exhibit

Which three header fields do not change for the duration of this call using SIP? (Choose three.) A. From tag B. To tag C. Contact: D. transaction ID in the Viheader E. request URI F. Call-ID: Ans: A, B, F

Q134. Refer to Exhibit.

Which data rate, in bits per second, would be negotiated for a T.38 fax call? A. 33600 B. 28800 C. 14400 D. 24000 E. 19200 Ans: C
The voice codec is G.711, fax transmission occurs at up to 14400 bps because 14400 bps is less than the 64-kbps voice rate. If the voice codec is G.729 (8 kbps), the fax transmission speed is 7200 bps.

Q135. Which 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. Ans: B, C, E
Router(conf-voi-serv)# modem passthrough nse payload-type 101 codec g711ulaw redundancy maximum-sessions 1

Q136. Refer to the exhibit.

What is the correct duration, in milliseconds, of the DTMF digit that is received? A. 30 B. 40 C. 60 D. 65 E. 101 Ans: C Q137. Which three services must be activated on Cisco Unified Presence in order for presence and instant messaging to be functional? (Choose three.) A. Cisco Unified Presence SIP Proxy B. Cisco AXL Web Service C. Cisco Bulk Provisioning Service D. Cisco Unified Presence Engine E. Cisco Unified Presence Sync Agent F. Cisco Serviceability Reporter Ans: A, D, E
You must activate the following services: Cisco UP SIP Proxy, Cisco UP Presence Engine, Cisco UP Sync Agent, Cisco UP XCP Connection Manager, Cisco UP XCP Authentication Manager

Q138. Which Cisco Unified Presence service parameter must be modified from the default value in order for presence and instant messaging to be functional? A. server name B. server IP address C. DNS domain D. SIP proxy domain E. enable presence Ans: D
services must be stopped on all nodes in the cluster before the Domain Name can be modified, then change FQDN to your email address FQDN: System > Cluster Topology > Settings > Domain Name:

Q139. What is required to back up Cisco Unified Presence configuration? A. tape backup device B. USB hard disk C. FTP server D. SFTP server E. TFTP server Ans: D

Q140. Refer to Exhibit.

Which of the certificates that are shown must be uploaded to Cisco Unified Presence when integrating the calendar with Exchange Server "email.cisco.com"? A. DST Root CA X3 only B. Cisco SSCA only C. email.cisco.com only D. DST Root CA X3 and Cisco SSCA E. Cisco SSCA and email.cisco.com Ans: D Q141. Which of these best describes the "Incoming ACL" configuration on Cisco Unified Presence? A. permits incoming packets to Cisco Unified Presence B. bypasses digest authentication C. allows instant messages D. allows incoming certificates to Cisco Unified Presence E. filters incoming presence status requests Ans: B In the Incoming and Outgoing Access Control List (ACL), you can configure patterns that control which incoming hosts and domains can access Cisco Unified Presence without authentication.

Q142. Which Cisco tool can be used to capture packets on Cisco Unified Presence? A. Cisco Unified Presence CLI B. System Troubleshooter on the Cisco Unified Presence web portal C. Cisco Unified RTMT D. Cisco Unified Presence "Cisco Unified Serviceability" web portal E. Cisco Unified Presence "Cisco Unified OS Administration" web portal Ans: A
utils network capture eth0

Q143. Which two Cisco Unified Contact Center Express system components do not support integration redundancy? (Choose two.) A. CTI ports B. AXL service C. Cisco Unified CM Telephony trigger D. CSQ E. dialog groups F. HTTP trigger Ans: D, F

Q144. Refer to Exhibit.

User "jdoe" was not able to download voice mail with his Cisco Unified Personal Communicator. Which configuration change on the Voicemail Profile Configuration page on Cisco Unified Presence is most likely to solve 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. Ans: D

Q145. Which three statements about the Outbound Dialer solution on Cisco Unified Contact Center Express are correct? (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. Ans: B, D, E
Do Not Call list. The United States Do Not Call list is not supported and must be filtered out manually before uploading. You must use your own tools to expunge Do Not Call numbers from their Contact List table. High availability requirements. For high availability deployments, all Unified CCX and database nodes must already be in service. IP Phone Agent. The IP Phone Agent is not supported

Q146. Which 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. Ans: A, E Q147. You have discovered that the Cisco Unified CM Telephony subsystem is in "PARTIAL_SERVICE" on a Cisco Unified Contact Center Express server. Which two misconfigurations could lead to this service state? (Choose two.) 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. Ans: B, E
I could see couple of reasons why it is out of service. Apr 13 10:43:54.108 AST %MIVR-SS_TEL-1-SS_OUT_OF_SERVICE:JTAPI subsystem in out of service: Failure reason=A number of route points are OOS - all routes; A number of CTI ports are OOS - all ports

Apr 13 10:43:54.108 AST %MIVR-SS_TEL-7-UNK:Number of CTI ports = 0 Action: 1. Please make sure that all your UCCX route points and CTI ports are in registered state in CUCM, by performing Data Resync operation (UCCX Admin->Subsystems->CM Telephony->Data Resync). 2. Also try creating a new call control group with few CTI ports and a test application with a route point and check is these parameters are successfully created.

Typical Unified CCX Call Flow

The basic Unified CCX call flow process is identified below: 1. A call arrives at voice gateway. 2. The voice gateway routes the call based on direction from the Unified CM (using H.323 or MGCP). 3. The Unified CM is configured for the dialed number to be routed by Unified CCX so a route request is sent to the Unified CCX server (using Unified CM Telephony). 4. Based upon the dialed number, the Unified CCX server selects an available CTI port and initiates the configured script. The first step in the work flow (accept) initiates the establishment of an Real-time Transport Protocol (RTP) VoIP data stream between the CTI port on the Unified CCX server and the VG port. In this scenario, we are assuming no appropriately skilled agents are available, so the application flow executes the queue loop logic until an agent becomes available. 5. An appropriately skilled agent becomes available. 6. The agent is selected/reserved by the Unified CCX server and this triggers the call to be transferred to the agent's phone and subsequently causes the agent's phone to ring (using Unified CM signaling). In addition, the Unified CCX server delivers a screen pop to the selected agent's desktop and enables the answer button on the agent's desktop. 7. The agent answers the call, which initiates the establishment of an RTP VoIP data stream between the agent's phone and the voice gateway port.

Q148. After logging into Cisco Agent Desktop, a Cisco Unified Contact Center Express agent could not go into ready state. 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. Ans: B, E Q149. Which 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 samples B. the time that is taken by the compression algorithm to correctly process sample block N C. the time that is taken to fill a packet payload with encoded/compressed speech D. the time that is required to clock a voice frame onto the network interface E. the time that is taken to queue a voice frame for transmission on the network connection Ans: C Packetization delay (n) is the time taken to fill a packet payload with encoded/compressed speech. Packetization delay can also be called Accumulation delay, as the voice samples accumulate in a buffer before they are released. Q150. Which 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 N B. the time that is taken by the DSP to compress a block of PCM samples C. the time that is taken to fill a packet payload with encoded/compressed speech D. the time that is required to clock a voice frame onto the network interface E. the time that is taken to queue a voice frame for transmission on the network connection Ans: B Coder delay is the time taken by the digital signal processor (DSP) to compress a block of PCM samples. This is also called processing delay (n). Q151. Which delay in a VoIP network is also known as accumulation delay? A. coder delay B. network switching delay C. queuing delay D. packetization delay E. dejitter delay Ans: D Q152. Which 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. Ans: D

This delay varies with the voice coder used and processor speed. For example, algebraic code excited linear prediction (ACELP) algorithms analyze a 10 ms block of PCM samples, and then compress them.

Q153. Which Cisco IOS command is used to define the size of the jitter buffer on Cisco IOS VoIP gateways? A. jitter-buffer B. expect-factor C. acc-qos D. playout-delay E. dejitter-buffer Ans: D The playout-delay command allows you to select a jitter buffer mode (fixed or adaptive) and specify certain values used by the DSP algorithms to adjust the size of the jitter buffer . Q154. Which three Cisco IOS commands can be used to verify configured playout delay values on Cisco VoIP gateways? (Choose three.) A. show voice call summary B. show call active voice C. show dial-peer voice tag number for dial peer D. show voice port voice interface number E. show voice dsp detail F. show voice accounting method Ans: B, C, D
Router# show dial-peer voice 302

Router# show call active voice Router# Show voice port 1/1/0
jitterOn voice calls, the packet inter-arrival variability caused by queuing delays and congestion in the network; the difference between when a packet is expected to arrive and when it actually is received. Jitter causes discontinuity in the real-time voice stream, which we hear as a choppy audio signal. It is a variable component of the total end-to-end network delay. jitter bufferStorage area used for handling voice packets that are in transit between the network and a codec. The jitter buffer's primary function is to reduce jitter by evening out the variability in the delivery times for voice packets.

Q155. Which 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 because of deep queues. Ans: A, D

Q156. Which 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. Ans: B, D Q157. CRTP belongs to which Cisco quality of service feature? A. classification B. congestion management C. congestion avoidance D. shaping and policing E. link efficiency mechanisms Ans: E
CRTP is supported on serial lines using Frame Relay, High-Level Data Link Control (HDLC), or PPP encapsulation. It is also supported over ISDN interfaces. You should configure CRTP if the following conditions exist in your network: Slow links and the need to save bandwidth

Q158. Refer to Exhibit.

Which statement about the QoS configuration for interface GigabitEthernet 1/0/1 on the Cisco Catalyst 3750 Series 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. Ans: D

Q159. Refer to Exhibit.

What is the correct expansion of the srr-queue abbreviation that is shown in the Cisco IOS command of the Catalyst 3750 Series Switch? A. shared round-robin queue B. serviced round-robin queue C. shaped round-robin queue D. special round-robin queue E. serial round-robin queue Ans: C Q160. Which statement about the Cisco Unity Connection message quota enforcement policies when a mailbox has exceeded the send/receive quota is incorrect? A. The user is unable to send messages. 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 non-delivery receipts to the senders. Ans: B

Q161. What is the default mailbox size that triggers disablement of sending and receiving voice messages for a Cisco Unity Connection user? A. 2 MB B. 4 MB C. 10 MB D. 14 MB E. 20 MB Ans: D

Q162. Which two statements about system broadcast messages on Cisco Unity Connection are correct? (Choose two.) 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. Ans: C, E System broadcast messages are played immediately after users log on to Cisco Unity Connection by phoneeven before they hear message counts for new and saved messages. After logging on, users hear how many system 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 montheven indefinitely. A user hears the system broadcast message the first time that he or she logs on to Connection during the period that the message is active. Users must listen to a system broadcast message in its entirety before Connection allows them to hear new and saved messages or to change setup options. Users cannot fast-forward or skip a system broadcast message. If a user hangs up before playing the entire system broadcast message, the message plays again the next time 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 or permanently 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 longer able 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 system broadcast messages when listening to messages in the Cisco Unity Inbox, an RSS reader, IMAP clients, Cisco Unified 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. Q163. Which two statements about system broadcast messages on Cisco Unity Connection are correct? (Choose two.) A. Users receive system broadcast messages even when they exceed their mailbox size limits and are no longer able to receive other messages. B. System broadcast messages trigger MWIs on user phones but do not trigger MWIs on alternate devices such 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. Ans: A, C

Q164. What is the maximum number of days for Cisco Unity Connection to retain expired system broadcast messages? A. 1 day B. 5 days C. 10 days D. 30 days E. 60 days Ans: E

Retention PeriodIndicates 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 date and time. To change the retention period for expired broadcast messages, enter a number from 1 to 60 days.

Q165. What is the default maximum recording length that is allowed for system broadcast messages on a Cisco Unity Connection server? A. 5 minutes B. 10 minutes C. 15 minutes D. 20 minutes E. 30 minutes Ans: A
Maximum Recording Lengthindicates the maximum length allowed for system broadcast messages. By default, senders can record messages up to 300,000 milliseconds (5 minutes) in length. To change the maximum recording length, enter a number from 60,000 (1 minute) to 36,000,000 (60 minutes) milliseconds.

Q166. What is the maximum recording length that is allowed for system broadcast messages on a Cisco Unity Connection server? A. 5 minutes B. 10 minutes C. 15 minutes D. 30 minutes E. 60 minutes Ans: E

Q167. Which of these is not a valid (Voice Profile for Internet Mail) VPIM message addressing option that is provided by Cisco Unity Connection to individuals on a remote voice messaging system? A. blind addressing B. Cisco Unity Connection directory C. implicit addressing D. private distribution list E. system distribution list Ans: C
Message Addressing Options : Connection directory, Blind addressing, Distribution lists

Q168. Which three call handlers are predefined on Cisco Unity Connection? (Choose three.) A. goodbye B. holiday greeting C. internal D. opening greeting E. operator F. closed G. standard Ans: A, D, E

Q169. Which greeting type is not a valid call handler on Cisco Unity Connection? A. busy B. closed C. external D. holiday E. standard Ans: C

Q170. Which two call handler greeting types on Cisco Unity Connection are overridden by the holiday greeting? (Choose two.) A. alternate B. busy C. closed D. error E. internal F. standard Ans: C, F

Q171. Which three call handler greeting types on Cisco Unity Connection are overridden by the internal greeting? (Choose three.) A. alternate B. busy C. closed D. error E. holiday F. standard Ans: C, E, F

Q172. Which two call handler greeting types on Cisco Unity Connection cannot be disabled? (Choose two.) A. alternate B. busy C. closed D. error E. holiday F. standard Ans: D, F Q173. Which three options are not valid application types on Cisco Unified Contact Center Express? (Choose three.) A. alternate application B. busy C. Cisco script application D. Cisco Unified CM Telephony E. Ring No Answer F. standard application Ans: A, D, F

Q174. What is the correct variable type for the "CSQ" variable in the "icd.aef" script on Cisco Unified Contact Center Express? A. string B. user C. queue D. document E. Boolean Ans: A Q175. What is the correct variable type for the "DelayWhileQueued" variable in the "icd.aef" script on Cisco Unified Contact Center Express? A. string B. user C. number D. integer E. Boolean Ans: D

Q176. Which statement about embedded Tcl scripts for B-ACD on Cisco Unified Communications Manager Express is correct? A. The Tcl scripts that are required for B-ACD services, along with the default audio files, must be available on the router flash memory. B. The Tcl scripts and the default audio files for B-ACD services are embedded natively in the Cisco IOS Software, 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 Ans: C http://blog.ipexpert.com/2009/01/24/b-acd-in-a-nutshell/

Q177. Which of these is not a mandatory configuration component to enable B-ACD service on Cisco Unified Communications Manager Express? A. an automated attendant Tcl script that handles the welcome prompt and menu choices B. a call-queue Tcl script that manages call routing and the queuing behavior number C. an ephone hunt group to receive calls from the call-queue service D. incoming dial peers for automated attendant pilot numbers E. Cisco Unity Express for voice mail to receive undelivered B-ACD calls Ans: E Q178. On Cisco Unified Communications Manager Express with a B-ACD application that is provisioned for four hunt groups--aa-hunt1, aa-hunt2, aa-hunt3, and aa-hunt4--which hunt group will be chosen when a caller dials 0? A. aa-hunt0 B. aa-hunt1 C. aa-hunt2 D. aa-hunt3 E. aa-hunt4 Ans: E Q179. What is the maximum number of calls that are allowed in each ephone hunt group call queue that is used by Cisco Unified Communications Manager Express B-ACD? A. 10 B. 15 C. 20 D. 25 E. 30 Ans: E

Q180. What is the default number of calls that are allowed in each ephone hunt group call queue that is used by Cisco Unified Communications Manager Express B-ACD? A. 5 B. 10 C. 15 D. 20 E. 30 Ans: B Q181. What is the maximum number of ephone hunt groups that can be used with a call-queue service by Cisco Unified Communications Manager Express B-ACD? A. 3 B. 5 C. 10 D. 15 E. 20 Ans: C Q182. Refer to Exhibit.

What is the pilot number for the ephone hunt group that is configured on this Cisco Unified Communications Manager Express with B-ACD? A. 1 B. 30 C. 3000 D. 3333 E. 5553000 Ans: C

Q183.

Which statement about the B-ACD configuration on Cisco Unified Communications Manager Express is correct? 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. Ans: D

Q184. Which attribute is not associable with a device profile on Cisco Unified Communications Manager? A. User Hold MOH Audio Source B. phone button template C. softkey template D. directory URL E. expansion module information Ans: D Q185. Which two attributes are associable with a device profile on Cisco Unified Communications Manager? (Choose two.) A. MLPP information B. Network Hold MOH Audio Source C. privacy D. directory URL E. authentication service URL Ans: A, C Q186. The Cisco Dialed Number Analyzer service belongs to which feature service group on Cisco Unified Communications Manager? A. Database and Admin Services B. Performance and Monitoring Services C. CM Services D. CTI Services E. Voice Quality Reporter Services Ans: C
Step 1 Access Cisco Unified Communications Manager Serviceability. Step 2 Choose Tools > Service Activation. The Service Activation window displays. Step 3 Select Cisco Dialed Number Analyzer from the Unified CMServices list and click Save.

Q187. The Cisco AXL Web Service belongs to which feature service group on Cisco Unified Communications Manager? A. Database and Admin Services B. Performance and Monitoring Services C. CM Services D. CTI Services E. Voice Quality Reporter Services Ans: A Q188. The Cisco Unified Communications Manager Assistant service belongs to which feature service group on Cisco Unified Communications Manager? A. Database and Admin Services B. Performance and Monitoring Services C. CM Services

D. CTI Services E. Voice Quality Reporter Services Ans: D

Q189. Which two of these are valid modes of operation for the Cisco Unified Communications Manager Assistant feature? (Choose two.) A. forwarded line support B. pickup line support C. proxy line support D. hybrid line support E. shared line support F. dual line support Ans: C, E

Q190. Which three features override the DND setting on an SCCP-controlled IP phone on Cisco Unified Communications Manager? (Choose three.) A. park reversion for remotely parked calls B. callback--terminating side C. MLPP D. hold reversion E. park reversion for locally parked calls F. remotely placed pickup request Ans: C, D, E Q191. Which two features do not override the DND setting on an SCCP-controlled IP phone on Cisco Unified Communications Manager? (Choose two.) A. park reversion for remotely parked calls B. MLPP C. callback--terminating side D. hold reversion E. intercom F. park reversion for locally parked calls Ans: A, C

Q192. Which statement about whisper intercom implementation on Cisco Unified Communications Manager is correct? 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. Ans: A

When a phone user dials a whisper intercom line, the called phone automatically answers using speakerphone mode, providing a one-way voice path from the caller to the called party, regardless of whether the called party is busy or idle. Unlike the standard intercom feature, this feature allows an intercom call to a busy extension. The calling party can 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 page and Cisco Unified CME plays a zipzip tone before the called party hears the caller's voice. If the called party wants to speak to the caller, the called party selects the intercom line button on their phone. The lamp for intercom buttons are colored amber to indicate one-way audio for whisper intercom and green to indicate twoway audio for standard intercom. You must configure a whisper intercom directory number for each phone that requires the Whisper Intercom feature. A whisper intercom directory number can place calls only to another whisper intercom directory number. Calls between a whisper intercom directory number and a standard directory number or intercom directory number are rejected with a busy tone.

Q193. Refer to Exhibit.

Which two statements about the "Operational VLAN Id" parameter on the Cisco IP phone Network Configuration menu are true? (Choose two.) A. This parameter can be configured from the Cisco Unified Communications Manager Web Administration Phone Configuration page. B. This parameter can be manually administered from the phone, as long as the Settings menu of the phone is unlocked. 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. Ans: C, D Q194. Refer to Exhibit.

Which two statements about the "Admin. VLAN Id" parameter on the Cisco IP phone Network Configuration menu are true? (Choose two.) A. This parameter can be configured from the Cisco Unified Communications Manager Web Administration Phone Configuration page. B. This parameter can be manually administered from the phone, as long as the Settings menu of the phone is unlocked. 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. Ans: B, C Q195. Which two strings are valid route patterns on Cisco Unified Communications Manager? (Choose two.) A. 123+ B. 123$ C. 123/ D. 123% E. 123D F. 123T Ans: A, E

Q196. All incoming calls to the Cisco Unified Communications Manager Express B-ACD are disconnected immediately. 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. Ans: D

I hope you guys like my explanatory notes for the above questions. Jaleel251@yahoo.com