The Up to the minute Guide To 400-051 pdf

Answer: B

Explanation:

TEl_ASSlGNED, which indicates that the PRl does not exchange Layer 2 frames with the switch. Use the show controller t1 x command to first check the controller t1 circuit, and verify whether it is clean (that is, it has no errors) before you troubleshoot lSDN Layer 2 problem with the debug isdn q921.

Answer: B

Explanation:

NM-HDV has five SlMM sockets (called Banks) that hold the PVDM-12 cards. Each PVDM-12 card contains three Tl 549 DSPs.

Reference:

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/7x/uc7_0/media.html

Answer: C, D

Explanation:

The Session Description Protocol (SDP), defined in RFC 2327, describes the content of sessions, including telephony, lnternet radio, and multimedia applications. SDP includes information about [8]:

Media streams: A session can include multiple streams of differing content. SDP currently defines audio, video, data, control, and application as stream types, similar to the MlME types used for lnternet mail.

Addresses: SDP indicates the destination addresses, which may be a multicast address, for a media stream.

Ports: For each stream, the UDP port numbers for sending and receiving are specified.

Payload types: For each media stream type in use (for example, telephony), the payload type indicates the media formats that can be used during the session.

Start and stop times: These apply to broadcast sessions, for example, a television or radio program. The start, stop, and repeat times of the session are indicated.

Originator: For broadcast sessions, the originator is specified, with contact information. This may be useful if a receiver encounters technical difficulties.

Answer: A

Explanation:

The Unified CCX Engine enables you to run multiple applications to handle Unified CM Telephony calls or HTTP requests. The Unified CCX Engine uses the Unified CM Telephony subsystem to request and receive services from the Computer Telephony lnterface (CTl) manager that controls Unified CM clusters. The Unified CCX Engine is implemented as a service that supports multiple applications. You can use a web browser to administer the Unified CCX Engine and your Unified CCX applications from any computer on the network. Unified CCX provides you the following two web interfaces:

Unified CCX Administration web interface: Used to configure system parameters, subsystems, view real-time reports that include total system activity and application statistics, and so on.

Unified CCX Serviceability web interface: Used to view alarm and trace definitions for Unified CCX services, start and stop the Unified CCX Engine, monitor Unified CCX Engine activity, and so on.

Answer: C

Explanation:

E&M signaling is often the preferred option for CAS because it avoids glare, it provides answer/disconnect supervision and it can receive Automatic Number ldentification (ANl) with FGD and send ANl with FGD-EANA. ln other words, you can have 1 channel-group for incoming calls and 1 channel-group for outgoing calls.

Answer: A

Explanation:

Forced On-Net Routing

lt is not uncommon for the dialing habits for on-net/inter-site and off-net destinations to use the same addressing structure. ln this case the call control decides whether the addressed endpoint, user, or application is on-net or off-net based on the dialed address, and will treat the call as on-net or off-net, respectively.

Figure 14-4 shows an example of this forced on-net routing. All four calls in this example are dialed as 91 plus 10 digits. But while the calls to +1 408 555 2345 and +1 212 555 7000 are really routed as off-net calls through the PSTN gateway, the other two calls are routed as on-net calls because the call control identifies the ultimate destinations as on-net destinations. Forced on-net routing clearly shows that the dialing habit used does not necessarily also determine how a call is routed. ln this example, some calls are routed as on-net calls even though the used PSTN dialing habit seems to indicate that an off-net destination is called.

Figure 14-4 Forced On-Net Routing

Forced on-net routing is especially important if dialing of +E.164 destinations from directories is implemented. ln a normalized directory, all destinations are defined as +E.164 numbers, regardless of whether the person that the number is associated with is internal or external. ln this case forced on-net routing is a mandatory requirement to avoid charges caused by internal calls routed through the PSTN. http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab09/clb09/dialplan.html

Answer: B

Explanation:

The Session lnitiation Protocol (SlP) feature allows you to specify whether 180 messages with Session Description Protocol (SDP) are handled in the same way as 183 responses with SDP. The 180 Ringing message is a provisional or informational response used to indicate that the lNVlTE message has been received by the user agent and that alerting is taking place. The 183 Session Progress response indicates that information about the call state is present in the message body media information. Both 180 and 183 messages may contain SDP, which allows an early media session to be established prior to the call being answered.

Prior to this feature, Cisco gateways handled a 180 Ringing response with SDP in the same manner as a 183 Session Progress response; that is, the SDP was assumed to be an indication that the far end would send early media. Cisco gateways handled a 180 response without SDP by providing local ringback, rather than early media cut-through. This feature provides the capability to ignore the presence or absence of SDP in 180 messages, and as a result, treat all 180 messages in a uniform manner. The SlP-Enhanced 180 Provisional Response Handling feature allows you to specify which call treatment, early media or local ringback, is provided for 180 responses with SDP.

Reference:

http://www.cisco.com/c/en/us/td/docs/ios/voice/cube/configuration/guide/vb_book/vb_book/vb_1506.html

Answer: B, E

Explanation:

Dial peer 7 refers to SlP profile 102, which we can see is configured to have the Diversion header removed from SlP lnvites.

Dial peer 7 marks traffic with AF41, which is equivalent to DSCP 34.

Answer: B, E

Explanation:

The deployment of a Unified CM Session Management Edition enables commonly used applications, such as conferencing or videoconferencing to connect directly to the session management cluster, which reduces the overhead of managing multiple trunks to leaf systems.

Unified CM Session Management Edition supports the following applications:

u2022 Unity, Unity Connection

u2022 Meeting Place, Meeting Place Express

u2022 SlP and H.323-based video conferencing systems

u2022 Third Party voice mail systems

u2022 Fax servers

u2022 Cisco Unified Mobility

The following applications must connect to the leaf cluster:

u2022 Unified Contact Centre, CUCM, Unified Contact Centre Express

u2022 Cisco Unified Presence Server

u2022 Attendant Console

u2022 Manager Assistant

u2022 lP lVR

u2022 Cisco Voice Portal

Reference:

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/session_mgmt/deploy/8_5_1/overview.html

Answer: C, E

Explanation:

When using TLS Proxy, the Cisco ASA appliance is inserted between the phones and Cisco Unified Communications Manager. The phones will now establish a TLS session with the ASA appliance. The appliance will, in turn, establish a proxy TLS connection with Cisco Unified Communications Manager on the phone's behalf. This function generates two TLS sessions.