GE0-807 実際試験問題と解答 - 模擬練習 - 受験自信を増やす

Question No : 1
You would like to record what the customer hears and says from the very beginning of the call to when it is terminated, regardless of whether or not an agent transferred the call, or created a conference, and so on.
Which ON type should you use to enable recording?

A.Extension
B.RP
C.Trunk
D.VOIP port

正解:

Question No : 2
Which of the following choices below contains all the messages that SIP is required to act on during the request/response handshake?

A.INVITE, 200 OK and ACK
B.INVITE, 200 O
C.ACK and 200 ACCEPTED
D.INVIT
E.200 OK
F.INVITE, 200 OK and 200 ACCEPTED

正解:
Explanation:
Reference: http://www.brocade.com/downloads/documents/html_product_manuals/SI_12502_ASLBG/wwhelp/wwhimpl/common/html/wwhelp.htm#context=SI_12502_ASLB_Guide&file=sip.3.2. html

Question No : 3
In a single site deployment, SIP Server can be deployed in which of the following modes? Choose 3 answers

A.Stand alone SIP Server
B.SIP Server integrated with a Softswitch
C.Open Standard SIP Architecture
D.Hybrid architecture
E.Application Server Mode

正解:
Explanation:
Reference: http://www.genesys.com/resources/training/certification/Exam_Study_Guide_GCP8_SIP.pdf (See the Describe SIP Server Deployment modes).

Question No : 4
Which of the following are SIP methods that can be used to pass DTMF tones? (Choose 3 answers)

A.In-band method by encoding DTMF tone as regular RTP packets
B.In-band method by encoding DTMF tone as in specific RTP packets (RFC 2833)
C.Out-of-band method by encoding DTMF tone in SIP message (UPDATE method)
D.Out-of-band method by encoding DTMF tone in SIP message (MESSAGE method)
E.Out-of-band method by encoding DTMF tone in SIP message (INFO method)

正解:
Explanation:
Reference: http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/sip/configuration/124t/sip-12-4t-book/voi-sip-dtmf.html#GUID-0E228A53-4405-4730-8D54-B37979E69A8B

Question No : 5
Which header in the INVITE message will SIP Server analyze in order to select the proper geo-location for an inbound call?

A.From
B.To
C.via
D.contact

正解:
Explanation:
Reference: http://docs.genesys.com/Special:Repository/sip81rn.html?id=c4d480e4-94a24cf8-ac24-5ccf3ccb7759

Question No : 6
If multiple patterns are matched in a dial plan, which of the following describes how SIP Server could determine which pattern match to use? (Choose 2 answers)

A.First Valid match
B.Lowest Dial-plan-rule-(number)
C.Most specific match
D.Last Valid match

正解:

Question No : 7
Which statements are correct? (Choose 2 answers)

A.A third-party Softswitch must be integrated with sip server using a CTI link.
B.If there are multiple outbound dialing rules configured within a single Class of Service (COS), and more than one rule matches the dialed number (containing a similar pattern). SIP Server will choose the outbound dialing rule with the most matched digits in it.
C.When integrating SIP Server with an approved third-party Softswitch, all SIP endpoints must be registered with the particular Softswitch and not the SIP server.
D.It is possible to configure Auto Agent Logout after a specific period of Agent inactivity on SIP Server 8.

正解:

Question No : 8
In the case where MSML is configured between SIP Server and Media Server, what is the SIP method used to carry information between them both?

A.XML
B.OPTION
C.T-Lib (Events/Requests)
D.voice XML
E.INFO
F.MESSAGE

正解:
Explanation:
Reference: https://tools.ietf.org/html/rfc5707 (See SIP INFO).

Question No : 9
Which protocol is SIP Server using in communication with Media Server?

A.CTI protocol
B.T-Lib
C.SIP
D.323
E.voice XML

正解:
Explanation:
Reference: http://en.wikipedia.org/wiki/List_of_SIP_software

Question No : 10
Which protocol is SIP Server using when in communication with Softswitch?

A.CTI protocol
B.T-Lib
C.SIP
D.Proprietary protocol

正解:

Question No : 11
Which of the following is SIP Server deployment modes used with a Framework 8 solution? Choose 3 answers

A.Stand Alone Mode
B.Application Server Mode
C.Multimedia Mode
D.Customer Side Proxy Mode
E.CTI Mode

正解:
Explanation:
Reference: http://genesyslab.info/wiki/index.php/Special:Repository/81fr_depsip.pdf?id=2e30d00a-05d6-4c84-a539-eb7ddcbde5f4 (page 34)

Question No : 12
Which of the following are valid parameters of the Announcement Treatment? (Choose 2 answers)

A.Language
B.ID
C.Music_DN
D.Duration

正解:
Explanation:
Reference: http://www.genesyslab.info/repository/PSDK/8.0Java_API_Reference/com/genesyslab/platform/voice/protocol/docfiles/TLib%20Datatypes/TTreatmentType.html (See the value pairs for treatment text to speech).

Question No : 13
What does 3PCC mean?

A.All call handling actions (answer, hold, initiate transfer, etc.) are performed on the SIP endpoint
B.All call handling actions (answer, hold, initiate transfer, etc.) are performed on the multiple PCs (In this case 3 PCs)
C.T-lib signaling takes over SIP signaling, so third party application such as Agent Desktop can transfer VoIP packets to SIP Server and back using T-lib
D.A T-lib client such as Agent Desktop can control a call which is actually between other parties such as SIP endpoints

正解:

Question No : 14
Which of the features below does a B2BUA deliver to a SIP-based VoIP architecture? (Choose 3 answers)

A.centralized call management
B.SIP-based VoIP interworking between LAN and WAN
C.management and monitoring of the entire call state
D.Provide conversion of TDM to DIP
E.connect legacy phones

正解:
Explanation:
Reference: http://www.eetimes.com/document.asp?doc_id=1203045

Question No : 15
Which of the following methods does SIP Server use to create a new SIP dialog? (Choose 2 answers)

A.INVITE
B.re-INVITE
C.REFER
D.REGISTER

正解:
Explanation:
Reference: https://tools.ietf.org/html/rfc5057

Genesys GE0-807 問題練習