[Nov-2021 Newly Released] Pass 300-815 Exam - Real Questions & Answers
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NEW QUESTION 27
Which call pickup feature allows users to pick up incoming calls in a group that is associated with their own group?
- A. Group Call Pickup
- B. BLF Call Pickup
- C. Other Group Pickup
- D. Directed Call Pickup
Answer: C
Explanation:
Reference:
https://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/4_1_3/ccmsys/ a07cpick.html#wp1022865
NEW QUESTION 28
Which two extended capabilities must be configured on dial peers for fast start-to-early media scenarios (H.323 to SIP interworking)? (Choose two.)
- A. DTMF
- B. AUDIO
- C. FAX
- D. VIDEO
- E. BFCP
Answer: A,E
Explanation:
Section: Signaling and Media Protocols
NEW QUESTION 29
Refer to the exhibit.
While troubleshooting call failures on the Cisco Unified Border Element, an administrator notices that messages are being sent to the service provider, but there is no response The administrator later learns that this SIP provider does not support PRACK. Which header should be removed from the SIP message to resolve this issue?
- A. Contact <sip:[email protected]:5060>
- B. Content-Type: application/sdp
- C. Content-Disposition: session:handling=required
- D. Require 100rel
Answer: D
NEW QUESTION 30
Refer to the exhibit.
Outbound calls to the service provider cause intermittent errors due to a codec mismatch. The internal network sends early offer SDP that contains only G.711 A-law. The service provider reports that some destinations support only G.711 A-law while others support only iLBC. The service provider also allows only 20 active calls at a time Which configuration allows successful media negotiation for all calls using outbound dial peers 5002 and 5003?
- A. Option A
- B. Option C
- C. Option B
- D. Option D
Answer: D
NEW QUESTION 31
Refer to the exhibit.
An administrator is troubleshooting a problem in which some outbound calls from an internal network to the Internet telephony service provider are not getting connected, but some others connect successfully. The firewall team found that some call attempts on port 5060 came from an unrecognized IP that has not been defined in the firewall rule. What should the administrator configure in the Cisco Unified Border Element to fix this issue?
- A. bind signaling and media to the loopback interface
- B. ip prefix-list to filter the unwanted IP address
- C. use of port 5061 for SIP secure
- D. access list allowing the firewall IP
Answer: A
NEW QUESTION 32
End users at a new site report being unable to hear the remote party when calling or being called by users at headquarters. Calls to and from the PSTN work as expected. To investigate the SIP signaling to troubleshoot the problem, which field can provide a hint for troubleshooting?
- A. c= line of SDP content
- B. Contact: header of the 200 OK response
- C. o= line of SDP content
- D. Allow: header if the 200 OK response
Answer: A
NEW QUESTION 33
Refer to the exhibit.
For long-distance calls, users must prefix their dialed number with "91." The translation pattern was created to strip the 91 as the PSTN expects a 10- digit number. The PSTN also requires the calling number to be set to 9195551234. However, the service provider has said calls with a different calling number are being received. How is this issue resolved?
- A. Disable Use Calling Party's External Phone Number Mask on the route pattern.
- B. Enable Force Authorization Code on the route pattern.
- C. Change the partition of the translation pattern from none to pstn_pt.
- D. Enable Use Calling Party's External Phone Number Mask on the translation pattern.
Answer: A
NEW QUESTION 34
After configuring a Cisco CallManager Express with Cisco Unity Express, inbound calls from the PSTN SIP trunk receive a ring tone for 20 seconds and then a busy signal instead of voicemail. Which configuration fixes this problem?
- A. Router(config)#dial-peer voice 2 voip
Router(config-dial-peer)#no vad - B. Router(config)# voice service voip
Router(conf-voi-serv)#no supplementary-service sip moved-temporarily - C. Router(config)# voice service voip
Router(conf-voi-serv)#allow-connections h323 to h323 - D. Router(config)# voice service voip
Router(conf-voi-serv)#allow-connections voice-mail mod
Answer: C
Explanation:
Section: Call Control and Dial Planning
Explanation/Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cusrst/admin/sccp_sip_srst/configuration/ guide/SCCP_and_SIP_SRST_Admin_Guide/srst_call_handling.html
NEW QUESTION 35
Refer to the exhibit.
A company needs to ensure that all calls are normalized to E164 format. Which configuration will ensure that the resulting digit string + 14085554001 is created and will be routed to the E.164 routing schema?
- A. Calling Party Transformation Mask of +14085554XXX
- B. Called Party Transformation Mask of + 1408555[35)XXX
- C. Calling Party Transformation Mask of +1408555XXXX
- D. Called Party Transformation Mask of + 14085554XXX
Answer: D
NEW QUESTION 36
Why would RTP traffic that is sent from the originating endpoint fail to be received on the far endpoint?
- A. The far end connection data (c=) in the SDP was overwritten by deep packet inspection in the call signaling path.
- B. The RTP traffic is arriving beyond the jitter buffer on the receiving end.
- C. A firewall in the media path is blocking TCP ports 16384-32768.
- D. Cisco Unified Communications Manager invoked media termination point resources.
Answer: B
NEW QUESTION 37
The administrator of ABC company is troubleshooting a one-way audio issue for a call that uses H.323 protocol (slow-start mode). The administrator requests that you provide the IP and port information of the Real- Time Transport Protocol traffic that had the one-way audio call.
You gather the H.225 and H.245 messages for one of the one-way audio calls. Where can you find the RTP IP and port information for both sides? (Note: This call flow has not invoked any media resources like MTP or transcoders).
- A. H.245 Open Logical Channel
- B. H.245 Terminal Capability Set
- C. H.245 Open Logical Channel Ack
- D. H.225 Connect
Answer: A
Explanation:
Reference:
http://ccievoicehopeful.blogspot.com/2012/09/h323-notes.html
NEW QUESTION 38
Refer to the exhibit.
Users report that when they dial to Cisco Unity Connection from an external network, they cannot enter any digits. Assuming only in-band DTMF is supported, what is a reason for this malfunction?
- A. The negotiated RTP port is outside of the range described by RFC, so inband DTMFs do not work.
- B. No DTMF is negotiated.
- C. The rtpmap:0 value for the negotiated codec is marking DTMF as inactive.
- D. There is SIP Delayed Offer. DTMF is supported only in Early Offer.
Answer: B
NEW QUESTION 39
Which call pickup feature allows users to pick up incoming calls in a group that is associated with their own group?
- A. Group Call Pickup
- B. BLF Call Pickup
- C. Other Group Pickup
- D. Directed Call Pickup
Answer: C
NEW QUESTION 40
Which two descriptions of the Standard Local Route Group deployment are true? (Choose two.)
- A. chooses the route group that is configured under the device pool of the calling-party device
- B. can be associated under the route group
- C. can be assigned directly to the route pattern
- D. chooses the route group that is configured under the device pool of the called-party device
- E. can be associated only under the route list
Answer: D,E
NEW QUESTION 41
An administrator is configuring a cluster for ILS and wants to limit the amount of entities that Cisco Unified Communications Manager can write to the database for data that is learned through ILS. Which service parameter is used to adjust this limit?
- A. Imported Dial Plan Replication Database Object Lower Limit
- B. Global Data Service Parameter Limit
- C. ILS Max Number of Learned Objects in Database
- D. ILS Active Learned Object Upper Limit
Answer: C
Explanation:
Section: Cisco Unified CM Call Control Features
Explanation/Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/12_5_1SU1/systemConfig/ cucm_b_system-configuration-guide-1251su1/cucm_b_system-configuration-guide-
1251su1_restructured_chapter_0100011.html#CUCM_TK_I7C708C2_00
NEW QUESTION 42
Refer to the exhibit.
Users report that when they dial the emergency number 9911 from any internal phone, it takes a long time to connect with the emergency operator. Which action resolves this issue?
- A. Check the Urgent Priority check box under 9.911 pattern.
- B. Adjust the service parameter T204 timer to the desired value.
- C. Adjust the service parameter T302 timet to the desired value.
- D. Point the emergency pattern directly to the PSTN gateway.
Answer: A
NEW QUESTION 43
Drag and drop the commands from the bottom to the blanks in the code to implement a translation rule to allow only 11 digits to be received over a SIP trunk to a SIP provider. The Cisco UCM is currently sending calls to the Cisco Unified Border Element in E.
164 format. Not all options are used.
Answer:
Explanation:
NEW QUESTION 44
A customer is using a SIP trunk to route calls to ITSP to decrease the possibility of downtime, the customer invested in a failover device How does the customer ensure reachability to ITSP, so that if one device on ITSP fails, the calls will be routed to another device?
- A. Enable transmit security status on the SIP security profile
- B. Enable ANAT on the SIP profile.
- C. Monitor the link using network management toots, and if it fails, manually change the routing to another working device.
- D. Enable SIP Option Ping on the SIP profile.
Answer: D
NEW QUESTION 45
An engineer must route all SIP calls in the form of <user>@example.com to the SIP trunk gateway corporate local. Which two SIP route patterns can be used to accomplish this task? (Choose two.)
- A. [email protected]
- B. *.*
- C. *@example.com
- D. example.com
- E. gateway.corporate.local
Answer: B,C
NEW QUESTION 46 
Refer to the exhibit. A user reports that when they call a specific phone number, no one answers the call, but when they call from a mobile phone, the call is answered. The engineer troubleshooting the issue is expecting the far-end gateway to cut through audio on the 183 Session Progress SIP message. Which SIP Profile configuration element is necessary for the Cisco Unified Communications Manager to send acknowledgement of provisional responses?
- A. Accept Audio Codec Preferences in Received Offer must be set to On.
- B. On the SIP Profile, the configuration parameter SIP Rel1XX Options must be set to Send PRACK for all 1xx Messages.
- C. Early Offer for G Clear Calls must be enabled.
- D. Allow Passthrough of Configured Line Device Caller Information must be enabled.
Answer: B
Explanation:
Section: Signaling and Media Protocols
Explanation
NEW QUESTION 47
Refer to the exhibit.
Calls incoming from the provider are not working through newly set up Cisco Unified Border Element. Provider engineers get the 404 Not Found SIP message. Incoming calls are coming from the provider with called number "222333444" and Cisco Unified Communications Manager is expecting the called number to be delivered as "444333222". The administrator already verified that the IP address of the Cisco Unified CM is set up correctly and there are no dial peers configured other than those shown in the exhibit. Which action must the administrator take to fix the issue?
- A. Fix the voice translation-rule to match specifically number "222333444" and change it to "444333222".
- B. Change the destination-pattern on the outgoing dial peer to match "444333222".
- C. Create specific matching for "222333444" on the incoming dial peer.
- D. Set up translation-profile on the incoming dial peer to match incoming traffic.
Answer: D
NEW QUESTION 48
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