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QUESTION 401
A collaboration engineer is designing Cisco IM&P implementation to support instant messaging logging for compliance.
Which two external databases can be used to support that functionality? (Choose two.)
A. Oracle database
B. MySQL database
C. Microsoft SQL database
D. PostgreSQL database
E. Informix SQL database
Answer: AD
QUESTION 402
Refer to the exhibit. A cisco collaboration engineer is troubleshooting a gateway and gatekeeper problem and sees this output from a debug command.
Which two configuration can cause this problem? (Choose two)
A. The same zone prefix is configured in two different gatekeepers
B. The same H323-ID is configured in two different gateways
C. The same gw-type-prefix is configured in two different zone subnets IDs
D. The same zone subnet ID is configured in two different gatekeepers
E. The same E164-ID is configured in two different gateways
Answer: BE
Explanation:
This output from the debug h225 asn1 command shows a registration reject reason of duplicateAlias.
RAS INCOMING PDU ::=
value RasMessage ::= registrationReject :
{
requestSeqNum 24
protocolIdentifier { 0 0 8 2250 0 3 }
rejectReason duplicateAlias:
{
}
gatekeeperIdentifier {“gk”}
}
This is usually the result of the gateway registering a duplicate of an E164-ID or H323-ID: Another gateway has already been registered to the gatekeeper. If it is a duplicated E164-ID, change the destination pattern configured under a POTS dial-peer associated with an FXS port. If it is a duplicated H323-ID, change the gateway’s H.323 ID under the H.323 VoIP interface.
http://www.cisco.com/c/en/us/support/docs/voice/h323/22378-gk-reg-issues.html#rr1
QUESTION 403
The Cisco Unified Border Element is configured using high availability with the Hot Standby Routing Protocol. Which two pieces of information can be gathered about the calls traversing these border elements? (Choose two.)
A. The total number of calls is 150.
B. The number of non-native calls is 70.
C. The number of native calls is 50.
D. The number of calls preserved is 220.
E. The total number of active calls is 100.
Answer: AB
QUESTION 404
Which statement describes virtual SNR DN configuration and behaviour on a Cisco Communication Manager Express IOS router?
A. A virtual SNR DN is a DN that must be associated with multiple registered IP phones
B. Mid-calls on virtual SNR DN can be pulled back as soon as a phone becomes associated with the DN
C. SNR feature can only be invoked if the virtual SNR DN is associated with at least one registered IP phone
D. A call that is ringing a virtual SNR DN prior to its association with a registered phone, cannot be answered by the phone even after the association is made
E. Virtual SNR DN supports either SCCP or SIP IP phone DNs
Answer: D
QUESTION 405
A CUCM engineer has deployed Type B SIP Phones on a remote site and no SIP dial rules were deployed for these phones. How Will CUCM receive the DTMF after the phone goes off- hook and the button are pressed?
A. Each digit will be received by CUCM in a SIP NOTIFY message as soon as they are pressed
B. The first digit will be received in a sip invite and subsequent digits will be received using NOTIFY message as soon as they are pressed.
C. Each digit bill be received by CUCM in a SIP INVITE as soon as the dial soft key has been pressed.
D. All digits will be received by CUCM in a SIP INVITE as soon as the dial soft key has been pressed
Answer: A
Explanation:
Type-B IP telephones offer functionality based on the Key Press Markup Language (KPML) to report user key presses. Each one of the user input events will generate its own KPML- based message to Unified CM. From the standpoint of relaying each user action immediately to Unified CM, this mode of operation is very similar to that of phones running SCCP.
Every user key press triggers a SIP NOTIFY message to Unified CM to report a KPML event corresponding to the key pressed by the user. This messaging enables Unified CM’s digit analysis to recognize partial patterns as they are composed by the user and to provide the appropriate feedback, such as immediate reorder tone if an invalid number is being dialed.
In contrast to Type-A IP phones running SIP without dial rules, Type-B SIP phones have no Dial key to indicate the end of user input. A user dialing 1000 would be provided call progress indication (either ringback tone or reorder tone) after dialing the last 0 and without having to press the Dial key. This behavior is consistent with the user interface on phones running the SCCP protocol.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/5x/50dialpl.html#wp1090653
https://supportforums.cisco.com/document/87236/working-concept-sccp-sip-phones-and-dial-rules
QUESTION 406
The Video engineer wants to enable the LATM codec to allow video endpoint to communicate over audio With other IP devices Which two Characteristic should the voice engineer be aware of before enabling LATM on the Cisco Unified border element router? (Choose two)
A. Dual tone Multi-frequency interworking with LATM codec is not supported
B. Codec transcoding between LATM and other codecs is not supported
C. SIP UPDATE message outlined in RFC3311 is not supported
D. Box-to-Box High availability support feature is not supported
E. Configure LATM under a voice class or dial peer is not supported
F. Basic calls using flow-around or flow-through is not supported
Answer: AB
Explanation:
http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book.pdf
QUESTION 407
Refer to the exhibit. A collaboration engineer configured MVA for a company using an existing Cisco IOS voice gateway. When testing inbound calls it is found that they are all failing.
Which two sets of configuration changes fix this problem? (Choose two)
A. Dial-peer voice 4100 posts
service ccm
destination-pattern 4100$
B. Dial-peer voice 4101 voip
dtmf-relay h245-alphanumeric
session target ipv4:172.16.100.50
C. Dial-peer voice 4101 voip
dtmf-relay h245-alphanumeric
session target ipv4:172.16.100.50
codec g711ulaw
D. Dial-peer voice 4100 pots
service CCM
E. Dial-peer voice 4100 pots
service CCM
Incoming called-number .T
F. Dial-peer voice 4101 VOIP
dtmf-relay h245-signal
session target ipv4:172.16.100.50
codec g711alaw
Answer: CE
Explanation:
Please note that, current scenario we are talking only inbound calls.
http://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/200457-Configure-the-Mobile-Voice-Access-Featur.html#anc6
Configuration Example
Configure your H323 gateway, in this case 10.106.103.149 is the CUCM address.
application
service mva http://10.106.103.149:8080/ccmivr/pages/IVRMainpage.vxml
dial-peer voice 5050 pots
service mva
incoming called-number 5050
no digit-strip
direct-inward-dial
!
dial-peer voice 3001 voip
destination-pattern 5050
session target ipv4:10.106.103.149
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
QUESTION 408
Refer to the exhibit. A customer has two Cisco unified communication manager 9.X clusters that serve the same location. An engineer has attempted to set up Enhanced Location call admission control so that any call within a site between phones on the two cluster do not decrement the available bandwidth to and from that site. However, the real time monitoring tool currently shows bandwidth being used from the site to hub_none when a call is placed between phone at the site.
Which action must be taken to correct this situation?
A. The link between clusters must be a type of inter-cluster trunk instead of a SIP trunk
B. The hub_none location must have a link configuration to the phantom location
C. The device pool names must match between clusters
D. The hub_none location must have a link configured to the shadow location
E. The SIP trunks should be changed to use the shadow location
Answer: E
QUESTION 409
A user has reported that when trying to access Visual Voicemail the Following error is received:
“Unable to open application please try again later if it continues to fail contact your administrator’. The collaboration engineer working on the problem found the following on the phone logs CVMInstallerModule STATUS_install_cancelled STATUS_INSTALL) _ERROR [thread=installer MQThread][class=cip midp midletsuite installerModule][function=update status] Midlet install Canceled/ERROR…visual Voicemail
How can this issue be resolved?
A. Replace the sever name with the server IP on service URL field
B. Eliminate the space in the service Name field
C. Configure DNS on phone configuration so it can resolve server name
D. Check the Enable checkbox on IP phone service configuration
Answer: B
Explanation:
Looks like a simple error in phone service’s display name: Visual VoiceMail
It needs to be exactly VisualVoiceMail without spaces (delete the space in the service Name field).
QUESTION 410
Refer to the exhibit A. CUBE Cluster is working in HSRP box-to-box failover model. When the phone A calls Cisco WebEx meeting server to start a conference session, no DTMF tones are recognized. Which configuration change will fix this problem when configured on both CUBEs?
A. Voice-class sip asymmetric payload dtmf in dial-peer configuration
B. dtmf-relay rtp-nte digitdrop in the dial-peer configuration
C. Media flow-around under voice service voip configuration
D. Modem relay nse payload-type101 under global sip configuration
E. Asymmetric payload full configured under global sip configuration
Answer: E
QUESTION 411
Refer to the exhibit. A user is on an outbound call through a cisco Unified border Element gateway. When the user places the call on hold, the remote party hears silence. The cisco unified communication manager cluster is using multicast music on hold. The cisco unified border element gateway is on the same subnet as the Cisco unified communication manager cluster. Which two options will resolve this issue? (Choose two)
A. Media flow-through must be configured
B. CCM-manager music-on-hold should be removed from the configuration
C. The session transport UDP command must be configured
D. The Cisco unified border Element router must be set up for gateway-based MOH
E. The pass-thru content SDP command should be removed
Answer: AE
Explanation:
Configuring the media flow-around command is required for Session Description Protocol (SDP) pass-through. When flow-around is not configured, the flow-through mode of SDP pass-through will be functional.
·When the dial-peer media flow mode is asymmetrically configured, the default behavior is to fallback to SDP pass-through with flow-through.
SDP pass-through is addressed in two modes:
·Flow-through–Cisco UBE plays no role in the media negotiation, it blindly terminates and re-originates the RTP packets irrespective of the content type negotiated by both the ends. This supports address hiding and NAT traversal.
·Flow-around–Cisco UBE neither plays a part in media negotiation, nor does it terminate and re-originate media. Media negotiation and media exchange is completely end-to-end.
When SDP pass-through is enabled, some of interworking that the Cisco Unified Border Element currently performs cannot be activated. These features include:
·Delayed Offer to Early Offer Interworking
·Supplementary Services with triggered Invites
·DTMF Interworking scenarios
·Fax Interworking/QoS Negotiation
·Transcoding
QUESTION 412
An engineer wants to configure a Cisco IOS router to allow for up to four simultaneous conference in CME mode. Which configuration meets the requirement?
A. dspfram profile 1 conference
Codec G711ulaw
Maximum sessions 4
Maximum conference-participants 8
Telephony-service
Sdspfarm units 4
Sdspfarm tag 4 hwcfb
Conference hardware
B. dspfram profile 1 conference
Codec G711ulaw
Maximum conference-participants 8
Telephony-service
Sdspfarm units 4
Sdspfarm tag 4 hwcfb
Conference hardware
Max-conference-participants 8
C. dspfram profile 1 conference
Codec G711ulaw
Maximum sessions 4
Maximum conference-participants 8
Telephony-service
Sdspfarm units 1
Sdspfarm tag 1 hwcfb
Conference hardware
Max-conferences 2
D. dspfram profile 1 conference
Codec G711ulaw
Maximum sessions 4
Maximum conference-participants 8
Telephony-service
Sdspfarm units 1
Sdspfarm tag 1 hwcfb
Conference hardware
Max-conferences 4
Answer: D
QUESTION 413
A cisco unified CM cluster is being set up for call control discovery using the service advertising framework. An engineer discovers that patterns are not being learned by the cluster.
Which two items must be checked in an attempt to resolve the issue? (Choose two)
A. The CCD block patterns are not preventing remote patterns from being entered into the local cache.
B. The hosted DN group on the cluster matches the patterns that should be learned
C. The CCD advertising service is activated in cisco unified CM serviceability
D. A CCD route partition has been assigned for learned patterns
E. The CCD requesting service is activated in cisco unified CM serviceability
F. The SIP trunk is enabled for call control discovery
Answer: AF
Explanation:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_0_2/ccmfeat/fsgd-802-cm/fscallcontroldiscovery.html
After you configure call control discovery, you may block learned patterns that remote call-control entities send to the local Cisco Unified Communications Manager. (Call Routing > Call Control Discovery > Blocked Learned Patterns Ensure that the CCD block patterns are not preventing remote patterns from being entered into the local cache.
The local Cisco Unified Communications Manager cluster uses SAF-enabled trunks that are assigned to the CCD requesting service to route outbound calls to remote call-control entities that use the SAF network.
The Cisco Unified Communications Manager cluster advertises the SAF-enabled trunks that are assigned to the CCD advertising service along with the range of hosted DNs; therefore, when a user from a remote call-control entity makes an inbound call to a learned pattern on this Cisco Unified Communications Manager, this Cisco Unified Communications Manager receives the inbound call from this SAF-enabled trunk and routes the call to the correct DN.
Ensure that the Sip trunk is enabled for call control discover.
QUESTION 414
Refer to the exhibit. A customer is configuring CAR costing for calls. When the customer runs the costing reports, calls are not being tagged correctly.
Which two changes allow proper costing to be determined for these calls? (Choose two)
A. The toll free area code field must be updated to include all toll free area codes
B. A new local pattern must be added with the pattern “K!”
C. A new pattern must be added for the 914 and 625 area codes
D. The items are out of order and must be sorted with the most specific at the top
E. Overlapping area codes on the trunks must be removed
F. All external patterns must be change to include the outside access code
Answer: AB
Explanation:
Choose System > System Parameters > Dial Plan Configuration.
The Dial Plan Configuration window displays.
In the Toll Free Numbers field, enter the numbers in your dial plan that can be placed without a charge.
If the number of digits dialed equals 10 and the pattern is K! (more than one digit, in this case a 10-digit number that starts with a trunk code), the call gets classified as Local.
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/service/9_0/car/CUCM_BK_CB39F074_00_cdr-analysis-reporting-administration-90/CUCM_BK_CB39F074_00_cdr-analysis-and-reporting-administration_chapter_011101.html
QUESTION 415
Refer to the exhibit. A collaboration engineer is configuring dynamic call routing and DN learning between two cisco UCM and two cisco UCM express.
What two configuration tasks will support this? (Choose two)
A. CME B should be configured as service advertisement forwarder only
B. CME B should be configured as service advertisement client and forwarder
C. Router A and B should be configured to use the same autonomous system number
D. Router A and CME B should be configured to use the same autonomous system number
E. Router B and CME B should be configured to use the same autonomous system number
F. CME B should be configured as service advertisement client only.
Answer: BC
QUESTION 416
Refer to the exhibit. A jabber user reports that desktop mode does not work. Connection status shows the following message “Connection error. Ensure the server information in the phone services tab on the option window is correct. contact your system administrator for assistance.”
Which two actions will resolve the problem? (Choose two)
A. Change LDAP port from 389 to 636 in LDAP directory configuration
B. Restart DirSync service under unified serviceability
C. Change LDAP port from 389 to 443 in LDAP director configuration
D. Change LDAP port from 389 to 3268 in LDAP director configuration
E. Restart CTI manager service under unified serviceability
F. Restart TFTP service under Unified serviceability
Answer: DE
QUESTION 417
A cisco Unified CM user is set up with one remote destination profile that has two remote destination numbers. First destination number is the user’s mobile phone in country A and the Second is a mobile phone located in country B. All outbound calls are centralized from the gateway at country A. The user reports that inbound calls are properly routed to the mobile phone as long as the user is in country A. but inbound calls are not successfully routed to country B?
What could resolve this issue? (Choose two)
A. The enable mobile connect option must be selected under the user’s second remote destination number
B. The value of remote destination limits should be change to 2 instead of the default value of 4 under the end user page
C. The enable mobile voice access option must be selected under the end user page
D. The value of maximum wait time for desk pickup should be change 20000 instead of the default of 10000, under the end user page
E. The rerouting calling search space assigned to the user’s remote destination profile must have access to international calls
Answer: AE
QUESTION 418
Refer to the exhibit. A CUCM engineer is working with Globalization and localization on H323 gateway.
Which four configuration changes are needed to achieve the result on the exhibit? (Choose four)
A. Create as CSS and PT for calling party transformation pattern
B. Create a transformation profile and add 9011 in the international number prefix field
C. Assign a transformation profile in the incoming transformation profile setting in the E 164 transformation number prefix field
D. Assign the calling party transformation CSS to the device pools in the cluster
E. Uncheck the use device pool calling party transformation CSS on all the phones
F. Add 9011 in the International Number prefix field on the Incoming Calling Party Settings.
G. Create a Calling Party Transformation pattern with pattern 90ll.! and DDi Predot.
H. Assign the Calling Party Transformation CSS to the Calling Party Transformation CSS field in theH323 Gateway.
Answer: ABCD
QUESTION 419
Which two actions does the cisco Unified IP phone use the initial Trust list to perform? (Choose two)
A. Decrypt secure XML files
B. Encrypt RTP traffic for IP phone that are not register to the same call manager cluster
C. Download background image files
D. Authenticate their configuration file signature
E. Talk securely to CAPF which is a prerequisite to support configuration files encryption
Answer: DE
QUESTION 420
Refer to the exhibit. Which three events happen When Alice calls [email protected] and the URI lookup policy on the Cisco Unified CM server has been set to case insensitive? (Choose three)
A. The RTP server routes the call to [email protected] because remote URIs have priority
B. The RTP sever looks up to see if [email protected] is associated to a local number
C. The San Jose server calls [email protected] upon receiving the invite request
D. The San Jose server provide carol’s directory URI using ILS exchange
E. The RTP server sends the call to [email protected] because it has priority
F. The RTP server drops the call because it has two identical matches
Answer: BDE
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