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Q131. Which statement is true when considering a Cisco VoIP environment for regional configuration?
A. G.711 requires 128K of bandwidth per call.
B. G.729 requires 24K of bandwidth per call.
C. The default codec does not matter if you have defined a hardware MTP in your Cisco Unified Communications Manager environment.
D. To deploy a Cisco H.323 gatekeeper, you must configure MTP resources on the gatekeeper and only use G.711 between regions.
Answer: C
Q132. What component acts as a user agent for both ends of a SIP call while Cisco Unified SIP SRST is providing failover during a WAN outage?
A. B2BUA
B. SIP server
C. SIP proxy
D. SIP SRST router
E. SIP registrar
Answer: A
Q133. With Cisco Extension Mobility, when a user logs in to a phone type which has no user device profile, what will happen to the phone?
A. The phone takes on the default clusterwide device profile.
B. The phone creates a new device profile automatically.
C. The phone immediately logs the user off.
D. The phone crashes and reboots.
Answer: A
Q134. Refer to the exhibit.
When the user of a phone registered to the Cisco Unified Communications Manager places a call to 3001 when the SAF network is down, what happens?
A. The call fails.
B. The call is rerouted to the PSTN with the constructed PSTN number as +442288223001
C. The call is rerouted to the PSTN with the constructed PSTN number as 442288223001
D. The call is rerouted to the PSTN with the constructed PSTN number as 0002288223001
E. The call is rerouted to the PSTN with the constructed PSTN number as +0002288223001
Answer: A
Explanation:
Incorrect Answer: B, C, D When the SAF forwarder loses network connection with its call-control entity, the SAF forwarder withdraws those learned patterns that were published by the call control entity. In this case, CCD requesting service marks those learned patterns as unreachable via IP, and the calls get routed through the PSTN gateway. Link:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmfeat/fscallcontrol discovery.html
Q135. You recently implemented call redundancy at a new remote site, and users report that calls are dropped when the remote site supposedly is in SRST. Which two actions must you take to troubleshoot the problem? (Choose two.)
A. Confirm that SRST is configured on the voice gateway.
B. Confirm that the site has an SRST reference that is correctly associated with the Cisco Unified Communications Manager group.
C. Confirm that a calling search space is assigned to the voice gateway in Cisco Unified Communications Manager.
D. Confirm that the site devices are associated with a Cisco Unified Communications Manager group and that four Cisco Unified Communications Manager servers are available.
E. Check the Region settings in Cisco Unified Communications Manager.
F. Restart Cisco Unified Communications Manager services to confirm that the server is working correctly.
Answer: A,B
Q136. Refer to the exhibit.
What is the correct value to use for the "DSCP for TelePresence Calls" Cisco CallManager service parameter?
A. 28 (011100)
B. 34 (100010)
C. 41 (101001)
D. 46 (101110)
Answer: A
Q137. You are deploying a remote office setup that connects with Cisco Unity Communications Manager at a hub location. You have an available dedicated bandwidth of 20% from the 2-Mb/s WAN circuit for VoIP that supports a maximum of 17 calls. Which codec do you configure in Cisco Unity Communications Manager to achieve this?
A. G.722
B. G.711
C. G.729
D. iSAC
E. GSM-FR
F. iLBC
Answer: C
Q138. Refer to the exhibit.
How many calls are permitted by the RSVP configuration?
A. one G.711 call
B. two G.729 calls
C. one G.729 call and one G.711 call
D. eight G.729 calls
E. four G.729 calls
Answer: B
Explanation:
Incorrect Answer: A, C, D, E In performing location bandwidth calculations for purposes of call admission control, Cisco Unified Communications Manager assumes that each call stream consumes the following amount of bandwidth:
.G.711 call uses 80 kb/s.
.G.722 call uses 80 kb/s.
.G.723 call uses 24 kb/s.
.G.728 call uses 26.66 kb/s.
.G.729 call uses 24 kb/s.
Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a02cac.html #wpxref28640
Q139. Which trunk should you use in an H.323 gatekeeper-controlled network?
A. H.323
B. H.225
C. SIP
D. Intercluster
E. MGCP FXO trunk
F. MGCP T1/E1 trunk
Answer: B
Q140. Which process can localize a global E.164 with + prefix calling numbers for inbound calls to an IP phone so that users see the calling number in a local format?
A. Calling number localization is done using translation patterns.
B. Calling number localization is done using route patterns.
C. Calling number localization is done by configuring a calling party transformation CSS at the phone.
D. Calling number localization is done by configuring a calling party transformation CSS at the gateway.
E. Calling number localization is done by configuring the phone directory number in a localized format.
Answer: C