What component does Cisco Unified Communications Manager Express use to match outbound dial peers?
A. destination pattern
B. incoming called-number
C. calling number ANI
E. port or session target
Correct Answer: A Section: CUCME Explanation
When a router receives a voice call, it selects an outbound dial peer by comparing the called number (the full E.164 telephone number) in the call information with the number configured as the destination pattern for the POTS dial peer Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmedialp.html
A customer is in the planning stages of deploying a Cisco Unified Communications solution for their company. Previously, they were leasing a traditional PBX system from the telco and they have very little experience with voice. The customer wants to know what two signaling methods between the IP phone and the Cisco Unified Communications Manager Express are available for their use. (Choose two)
Correct Answer: BD Section: CUCM Explanation
Skinny Client Control Protocol (SCCP)
SCCP uses Cisco-proprietary messages to communicate between IP devices and Cisco CallManager. SCCP easily coexists in a multiple protocol environment. The Cisco IP Phone is an example of a device that registers and communicates with Cisco CallManager as an SCCP client. During registration, a Cisco IP phone receives its line and all other configurations from Cisco CallManager. Once it registers, it is notified of new incoming calls and can make outgoing calls. The SCCP protocol is used for VoIP call signaling and enhanced features such as Message Waiting Indication (MWI).
Session Initiation Protocol (SIP)
The Internet Engineering Task Force (IETF) developed the SIP standard for multimedia calls over IP. ASCII-based SIP works in client/server relationships as well as in peer-to-peer relationships. SIP uses requests and responses to establish, maintain, and terminate calls (or sessions) between two or more end points. Refer to the Understanding Session Initiation Protocol (SIP) chapter for more information on SIP and the interaction between SIP and Cisco Call Manager.
The International Telecommunications Union (ITU) developed the H.323 standard for multimedia communications over packet networks. As such, the H.323 protocol is a proven ITU standard and provides multivendor interoperability. The H.323 protocol specifies all aspects of multimedia application services, signaling, and session control over an underlying packet network. Audio is standard on H.323 networks, but networks can be scaled to include both video and data. The H.323 protocol can be implemented in large enterprise networks or can be deployed over an existing infrastructure, which makes H.323 an affordable solution.
The basic components of the H.323 protocol are terminals, gateways, and gatekeepers (which provide call control to H.323 endpoints). Similar to other protocols, H.323 applies to point-to-point or multipoint sessions. However, compared to MGCP, H.323 requires more configuration on the gateway.
Media Gateway Control Protocol (MGCP)
MGCP provides Cisco CallManager a powerful, flexible and scalable resource for call control. Cisco CallManager uses MGCP to control media on the telephony interfaces of a remote gateway and also uses MGCP to deliver messages from a remote gateway to appropriate devices.
MGCP enables a call agent (media gateway controller) to remotely control and manage voice and data communication devices at the edge of multiservice IP packet networks. Because of its centralized architecture, MGCP simplifies the configuration and administration of voice gateways and supports multiple (redundant) call agents in a network. MGCP does not provide security mechanisms such as message encryption or authentication.
Using MGCP, Cisco CallManager controls call processing and routing and provides supplementary services to the gateway. The MGCP gateway provides call preservation (the gateway maintains calls during failover and fallback), redundancy, dial-plan simplification (the gateway requires no dial-peer configuration), hookflash transfer, and tone on hold. MGCP-controlled gateways do not require a media termination point (MTP) to enable supplementary services such as hold, transfer, call pickup, and call park. If the MGCP gateway loses contact with its Cisco Call Manager, it falls back to using H.323 control to support basic call handling of FXS, FXO, T1 CAS, and T1/E1 PRI interfaces.
Which best describes the auto-attendant in a Cisco Unified Communications environment?
A. A set of call processing instructions that automatically tell the system what to do when it reaches a particular system ID
B. A system that automatically allows inside or outside callers to leave voice-mail messages 24 hours, 7 days a week, even when no operator is on duty
C. A function that greets and guides callers through a telephony system in a friendly and timely manner, allowing them to reach an endpoint, leave messages, or speak to an operator
D. A function that allows the option of listening to, composing, replying to, forwarding, or deleting calls or voice-mail messages through a website without the need of a live telephone operator
Correct Answer: C Section: CUCM Explanation
Cisco Unified Communications Manager Auto-Attendant, a simple automated attendant, allows callers to locate people in your organization without talking to a receptionist. You can customize the prompts that are played for the caller, but you cannot customize how the software interacts with the customer.
Which statement about the Cisco Unity Express default AutoAttendant is true?
A. The default AutoAttendant must be enabled during the Initialization Wizard process otherwise voice-mail services will not function.
B. Enabling the default AutoAttendant is not mandatory during the Initialization Wizard process.
C. The default AutoAttendant is enabled by default with the exception of the prompts that must be recorded via AvT.
D. The default AutoAttendant cannot be used as is, and it must be customized for the particular environment it will be used in.
Correct Answer: B Section: Unity Connection Explanation
The AutoAttendant can be configured at a later stage.
You have been tasked to configure a Cisco Unity Express system with a voicemail pilot number of 1900, an AutoAttendant with pilot number of 2900, and an Administration via Telephone pilot number of 3900. What is the minimum number of SIP dial peers required?
Correct Answer: A Section: Unity Connection Explanation
Ensurepass offers Latest 2013 640-461 Real Exam Questions , help you to pass exam 100%.