400-051 Preparation Kits

Author: Monika Bergmann

Question: 1

Refer to the exhibit.

The MGCP debugs were captured on a Cisco IOS MGCP PRI gateway registered to a Cisco Unified CM. Assume that this gateway had no active calls and will not take any new calls for the next 3 minutes. What time it will send the next NTFY message to the Cisco Unified CM?

A. Jan 10 05:56:35.130

B. Jan 10 05:55:45.130

C. Jan 10 05:55:50.130

D. Jan 10 05:56:05.130

E. Jan 10 05:55:40.130

Answer: C

Question: 2

DRAG DROP

An engineer is setting up a proxy TFTP between multiple Cisco communication Manager clusters.

Drag the step from the left to the correct order on the right to properly configure the certificates for the proxy TFTP. Not all options will be used.

Select and Place:

Answer:

Question: 3

Refer to the exhibit.

A cisco collaboration engineer discovers that an instance of IOS media termination point (MTP) could not maintain stable registration with CUCM. Call manager traces is showing in the exhibit. What is the reason for the flapping registration?

A. The CCM version on IOS configuration does not match the CUCM version.

B. The IOS MTP is experiencing high CPU and is missing its keep-alive.

C. A Firewall is blocking port 2000 intermittently between IOS Device and CUCM.

D. Another IOS Media device is attempting to register with the same name.

Answer: D

Question: 4

A CUCM engineer has deployed Type B SIP Phones on a remote site and no SIP dial rules were deployed for these phones. How Will CUCM receive the DTMF after the phone goes off- hook and the button are pressed?

A. Each digit will be received by CUCM in a SIP NOTIFY message as soon as they are pressed

B. The first digit will be received in a sip invite and subsequent digits will be received using NOTIFY message as soon as they are pressed.

C. Each digit bill be received by CUCM in a SIP INVITE as soon as the dial soft key has been pressed.

D. All digits will be received by CUCM in a SIP INVITE as soon as the dial soft key has been pressed

Answer: A

Explanation:

Type-B IP telephones offer functionality based on the Key Press Markup Language (KPML) to report user key presses. Each one of the user input events will generate its own KPML-based message to Unified CM. From the standpoint of relaying each user action immediately to Unified CM, this mode of operation is very similar to that of phones running SCCP.

Every user key press triggers a SIP NOTIFY message to Unified CM to report a KPML event corresponding to the key pressed by the user. This messaging enables Unified CM's digit analysis to recognize partial patterns as they are composed by the user and to provide the appropriate feedback, such as immediate reorder tone if an invalid number is being dialed.

In contrast to Type-A IP phones running SIP without dial rules, Type-B SIP phones have no Dial key to indicate the end of user input. A user dialing 1000 would be provided call progress indication (either ringback tone or reorder tone) after dialing the last 0 and without having to press the Dial key. This behavior is consistent with the user interface on phones running the SCCP protocol.

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/5x/50dialpl.html#wp1090653

https://supportforums.cisco.com/document/87236/working-concept-sccp-sip-phones-and-dial-rules

Question: 5

Which two Cisco Unified Communications Manager Express hunt group mechanisms keep track of the number of hops in call delivery decisions? (Choose two.)

A. sequential

B. peer

C. longest idle

D. parallel

E. overlay

F. linear

Answer: BC

Question: 6

Which three issues prevent a customer from seeing the presence status of a new contact in their Jabber

contact list? (Choose three.)

A. Incoming calling search space on SIP trunk to IM&P

B. IM&P incoming ACL blocking inbound status

C. Subscribe calling search space on SIP trunk to IM&P

D. PC cannot resolve the FQDN of IM&P

E. Owner user ID is not set on device

F. Primary DN is not set in end user configuration for that user

G. Subscriber calling search space is not defined on user's phone

Answer: BCD

No Presence Information After Login

Problem

You receive no Presence information after login.

Solution

Complete these steps in order to resolve this issue:

Make sure that the DNS server the PC is pointed to can resolve the fully qualified name of the CUPS server.

The host entry will not suffice, you must resolve via DNS.

Check the SUBSCRIBE CSS on the SIP trunk to CUP.

This CSS must include the partitions of the devices you are trying to receive status on.

The CUP SIP proxy incoming access control list (ACL) is not allowing incoming SIP presence messages to reach the presence engine. As a test, set the incoming ACL to ALL and reset the SIP proxy and presence engine. Log in again to the CUPC and try to reconfigure the incoming ACL properly.

http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-presence/97443-cups-cupc-ts.html

Question: 7

Refer to the exhibit.

An engineer is trying to provision CUCME with three 8841 phones. However all phone fail to register. Which two changes in the configuration would allow the phones to register? (Choose two)

A. The registrar server command must be added under the voice register global configuration

B. The IP address trusted authenticate command must be added under voice service voip

C. The source-address command must be added under the voice register global configuration

D. The local SIP proxy address must be configuration under the sip-ua configuration

E. The registrar server command must be added under the sip section of voice service voip

Answer: CE

Question: 8

A collaboration engineer has been asked to implement secure real-time protocol between a Cisco Unified CM and SIP gateway. Which option is a consideration for this implementation?

A. Only T.38 and Cisco fax protocol are supported

B. SIP require the all the time be sent in GMT

C. Call hold RE-INVITE is not supported

D. SRTP is supported only in cisco IOS 15.x and higher

Answer: B

As necessary, configure the router to use Greenwich Mean Time (GMT). SIP requires that all times be sent in GMT. SIP INVITE messages are sent in GMT. However, the default for routers is to use Coordinated Universal Time (UTC). To configure the router to use GMT, issue the clock timezone command in global configuration mode and specify GMT.

http://www.cisco.com/c/en/us/td/docs/ios/voice/cube/configuration/guide/vb_book/vb_book/vb_8240.html

Question: 9

Refer to the exhibit.

A collaboration engineer configures Cisco Unified CM location using G.711 and iLBC for each site. The bandwidth for each link is shown. Which two options represent the maximum concurrent number of calls supported from Grand Junction to Casper for each Codec? (Choose two)

A. 20 G.711 calls

B. 18 G.711 calls

C. 36 iLBC calls

D. 42 iLBC calls

E. 11 G.711 calls

F. 51 iLBC calls

Answer: CE

Question: 10

A collaboration engineer is troubleshooting an MOH problem on a Cisco IOS SIP gateway. While searching through a debug ccsip message output, which three parameters in the SIP messages can be used to determine if the call was placed on hold? (Choose three)

A. OPTIONS WITH 301 CALLHOLD

B. INVITE WITH a=INACTIVE

C. INVITE WITH a=SENDONLY

D. OPTION WITH c=INACTIVE

E. c=IN IP4 0.0.0.0

F. BYE WITH A = CALLHOLD

Answer: BCE

Explanation:

When a call is on hold you get something like this:

v=0

  • CiscoSystemsCCM-SIP 919861 2 IN IP4 10.10.30.14

s=SIP Call

c=IN IP4 0.0.0.0

t=0 0

m=audio 30120 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=ptime:20

a=inactive

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

The connection parameter shows 0.0.0.0, When the call is taking off hold you, the connection parameter should indicate the ip address where media is sent to. So it will have a real value. ( This is usual sent in the ACK.) cucm still sends a DO in the re-INVITE and the far end sends a 200 Ok with SDP. CUCM then sends ACK with SDP

This is how the whole call hold/transfer/call resume works

  1. CUCM sends invite (re-invite) with an inactive SDP (a=inactive) to indicate a break in media path
  2. CUCM sends a Delayed offer to insert MOH or to resume Media stream

NB: CUCM expects a sendrecv offer with SDP to the DO. (NB:if cucm gets an inactive offer SDP in the 200 OK instead of providing a send-recv offer SDP, the media path remains in an inactive state and causes calls to dropcall will drop),CUCM sends an ACK with sendonly to the 200 OK

  1. CUCM establishes newcall leg with the intended transfered destination..Once this call is connected
  2. CUCM receives a new Transfer instruction from the transferring phone to connect the held party
  3. Next CUCM sends a re-invite with an inactive SDP to indicate a break in media path to MOH (in attempt to complete transfer)
  4. Next CUCM sends an inactive SDP to indicate a break in media path between transfering party & transfered party to remove Xferring party from call
  5. Next CUCM sends a DO re-invite to connect the transfered party. The far end then sends 200 OK with the required SDP to connect the call