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Refer to the exhibit.
Which SIP response message should the TGW send if it cannot process the tunneled QSIG messages from the OGW?
- A. 405 Method Not Allowed
- B. 406 Not Acceptable
- C. 412 Conditional Request Failed
- D. 415 Unsupported Media Type
- E. 485 Ambiguous D
Answer : Explanation
Explanation:
Fallback from QSIG Tunneling -
In some situations, QSIG tunneling will fail or need to fall back: "¢ Remote party does not support multipart MIME body: In this case, the remote side sends a "415 Media Not Supported" response. Upon receiving this response,
OGW will fall back to normal mode and send an INVITE request without any tunneled content. This procedure helps ensure that at least the basic call will work normally. "¢ Remote party does not understand tunneled content: If the remote side does not support the tunneled content, it should drop the tunneled content and continue as a normal call; because all essential parameters are present in the original INVITE, the call can go through without the need for fallback.
References: http://www.cisco.com/c/en/us/solutions/collateral/enterprise-networks/empowered-branch-solution/white_paper_c11_459092.html
Refer to the exhibit.
During a QSIG tunneling over SIP call establishment, which two types of SIP messages can the OGW use to tunnel a waiting QSIG message? (Choose two.)
- A. SIP re-INVITE
- B. SIP NOTIFY
- C. SIP INFO
- D. SIP OPTIONS
- E. SIP UPDATE
- F. SIP REFER
Answer : AE
Explanation:
The TGW sends and the OGW receives a 200 OK response--the OGW sends an ACK message to the TGW and all successive messages during the session are encapsulated into the body of SIP INFO request messages. There are two exceptions:
When a SIP connection requires an extended handshake process, renegotiation, or an update, the gateway may encapsulate a waiting QSIG message into a SIP re-INVITE or SIP UPDATE message during QSIG call establishment.
When the session is terminated, gateways send a SIP BYE message. If the session is terminated by notice of a QSIG RELEASE COMPLETE message, that message can be encapsulated into the SIP BYE message.
References: http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/sip/configuration/15-mt/sip-config-15-mt-book/voi-sip-tdm.html
In a SIP REFER-based call transfer, which SIP message is being used by the recipient to notify the originator that the final recipient was successfully contacted?
- A. 200 OK
- B. NOTIFY with a message body of 200 OK
- C. 202 Accepted
- D. 100 Trying
- E. 200 BYE
Answer : B
Explanation:
The Refer method always begins within the context of an existing call and starts with the originator. The originator sends a Refer request to the recipient (user agent receiving the Refer request) to initiate a triggered Invite request. The triggered Invite request uses the SIP URL contained in the Refer-To header as the destination of the Invite request. The recipient then contacts the resource in the Refer-To header (final-recipient), and returns a SIP 202 (Accepted) response to the originator. The recipient also must notify the originator of the outcome of the Refer transaction--whether the final-recipient was successfully or unsuccessfully contacted. The notification is accomplished using the Notify Method, SIP"™s event notification mechanism. A Notify message with a message body of SIP 200
. If the call was successful, a call
OK indicates a successful transfer, while a body of SIP 503 Service Unavailable indicates an unsuccessful transfer between the recipient and the final-recipient results.
Refer to the exhibit.
Which user agent has the recipient role in this SIP REFER call transfer?
- A. user agent A
- B. user agent B
- C. user agent C
- D. user agent B and C
- E. user agent A and B
Answer : B
Explanation:
The Refer method has three main roles:
"¢ Originator""User agent that initiates the transfer or Refer request. "¢ Recipient""User agent that receives the Refer request and is transferred to the final-recipient. "¢ Final-Recipient""User agent introduced into a call with the recipient.
References: http://www.cisco.com/c/en/us/td/docs/ios_xr_sw/iosxr_r3-4/sbc/configuration/guide/sbc_c34/sbc34stx.pdf
Refer to the exhibit.
You received this debug output to troubleshoot a Cisco IOS MGCP gateway media-related problem at a customer site. What is the purpose of this message?
- A. The MGCP gateway is responding to an RQNT message from Cisco Unified Communications Manager to poll the media capabilities on its endpoints.
- B. The MGCP gateway is responding to an AUEP message from Cisco Unified Communications Manager to poll the media capabilities on its endpoints.
- C. The MGCP gateway is responding to an AUCX message from Cisco Unified Communications Manager to poll the active calls on its endpoints.
- D. The MGCP gateway is responding to an MDCX message from Cisco Unified Communications Manager during a call setup.
- E. The MGCP gateway is responding to a CRCX message from Cisco Unified Communications Manager during a call setup.
Answer : E
Explanation:
Which device is the initiator of a StationD message in a Cisco Unified Communications Manager SDI trace?
- A. SCCP IP phone
- B. SIP IP phone
- C. Cisco Unified Communications Manager
- D. MGCP analog gateway
- E. digital voice gateway
Answer : C
Explanation:
All messages to and from a skinny device are preceded by either the words StationD or StationInit. StationD messages are sent from call manager to IP phone.
Skinny message transmission such as this between the IP phone and CallManger occurs for every action undertaken by the IP phone, includinginitialization, registration, on-hook, off-hook, dialing of the digits, key press on the phone, and so much more.
What is the maximum length of any numeric geographic area address in ITU recommendation E.164?
- A. 15
- B. 18
- C. 21
- D. 22
- E. 25
Answer : A
Explanation:
E.164 defines a general format for international telephone numbers. Plan-conforming numbers are limited to a maximum of 15 digits. The presentation of numbers is usually prefixed with the character + (plus sign), indicating that the number includes the international country calling code (country code), and must typically be prefixed when dialing with the appropriate international call prefix, which is a trunk code to reach an international circuit from within the country of call origination.
According to ITU-T E.164 recommendations, which two fields in the National Significant Number code may be further subdivided? (Choose two.)
- A. Country Code
- B. National Destination Code
- C. Subscriber Number
- D. Regional Significant Number
- E. Local User Code
- F. National Numbering Plan
Answer : BC
Explanation:
A telephone number can have a maximum of 15 digits .The first part of the telephone number is the country code (one to three digits) .The second part is the national destination code (NDC). The last part is the subscriber number (SN). The NDC and SN together are collectively called the national (significant) number.
Refer to the exhibit.
A user is going through a series of dialing steps on a SIP Type B IP phone (for example, a Cisco 7975) to call an SCCP IP phone. Both phones are registered to the same Cisco Unified Communications Manager cluster. Assuming that the calling SIP phone is not associated with any SIP dial rules, which statement about how digits are forwarded to Cisco Unified Communications Manager for further call processing is true?
- A. Each digit is sent to Cisco Unified Communications Manager in a SIP NOTIFY message KPML event, at the time that the user enters the digit on the keypad.
- B. The SIP IP phone will wait for the interdigit timer to expire, or for the Dial softkey to be selected before sending each digit to Cisco Unified Communications Manager as a separate KPML event in a SIP NOTIFY message.
- C. The SIP IP phone will wait for the interdigit timer to expire, or for the Dial softkey to be selected before sending all digits to Cisco Unified Communications Manager in a SIP INVITE message.
- D. The SIP IP phone will wait for the interdigit timer to expire or for the Dial softkey to be selected before sending the first digit in a SIP INVITE and the subsequent digits in SIP INFORMATION messages.
- E. The SIP IP phone will send all digits to Cisco Unified Communications Manager in a SIP INVITE message as soon as the fourth digit is pressed. A
Answer : Explanation
Explanation:
KPML procedures use a SIP SUBSCRIBE message to register for DTMF digits. The digits themselves are delivered in NOTIFY messages containing an XML encoded body. And it is Out of Band DTMF
Refer to the exhibit.
A user is going through a series of dialing steps on a SIP Type B IP phone (for example, a Cisco 7975) to call an SCCP IP phone. Both phones are registered to the same Cisco Unified Communications Manager cluster. Assuming the calling SIP phone is associated with a SIP dial rule with a pattern value of 2001, which statement about how digits are forwarded to Cisco Unified Communications Manager for further call processing is true?
- A. As each digit is pressed on the SIP IP phone, it is sent to Cisco Unified Communications Manager in a SIP NOTIFY message as a KPML event.
- B. The SIP IP phone will wait for the interdigit timer to expire, and then send each digit to Cisco Unified Communications Manager as a separate KPML event in a SIP NOTIFY message.
- C. The SIP IP phone will wait for the interdigit timer to expire, or for the Dial softkey to be selected before sending all digits to Cisco Unified Communications Manager in a SIP INVITE message.
- D. The SIP IP phone will wait for the interdigit timer to expire, or for the Dial softkey to be selected before sending the first digit in a SIP INVITE and the subsequent digits in SIP INFORMATION messages.
- E. The SIP IP phone will wait for the interdigit timer to expire, and then send all digits to Cisco Unified Communications Manager in a SIP INVITE message.
Answer : E
Explanation:
Cisco Type B SIP Phones offer functionality based SIP INVITE Message. Every key the end user presses triggers an individual SIP message. The first event is communicated with a SIP INVITE, but subsequent messages use SIP NOTIFY messages. The SIP NOTIFY messages send KPML events corresponding to any buttons or soft keys pressed by the user. Cisco Type B SIP IP Phones with SIP dial rules operate in the same manner as Cisco Type A phones with dial rules.
Refer to the exhibit.
A user is going through a series of dialing steps on a SIP Type B IP phone (for example, a Cisco 7975) to call an SCCP IP phone. Both phones are registered to the same Cisco Unified Communications Manager cluster. Assuming the calling SIP phone is associated with a SIP Dial Rule with a pattern value of 2001, which statement about the call setup process of this call is true?
- A. Each digit will arrive at Cisco Unified Communications Manager in a SIP NOTIFY message as a KPML event, and Cisco Unified Communications Manager will extend the call as soon as the collected digits match the extension of the SCCP IP phone, bypassing class of service configuration on both IP phones.
- B. Each digit will arrive at Cisco Unified Communications Manager in a SIP NOTIFY message as a KPML event. When the collected digits match the extension of the SCCP IP phone, Cisco Unified Communications Manager will extend the call only if the class of service configuration on both phones permits this action.
- C. As soon as the user selects the Dial softkey, the SIP IP phone will forward all digits to Cisco Unified Communications Manager in a SIP INVITE message. Cisco Unified Communications Manager will extend the call as soon as the collected digits match the extension of the SCCP IP phone, bypassing class of service configuration on both IP phones.
- D. As soon as the user selects the Dial softkey, the SIP IP phone will forward all digits to Cisco Unified Communications Manager in a SIP INVITE message. Cisco Unified Communications Manager will extend the call only if class of service configuration on both phones permits this action.
- E. The SIP IP phone will wait for the interdigit timer to expire, and then send all digits to Cisco Unified Communications Manager in a SIP INVITE message. Cisco Unified Communications Manager will extend the call as soon as the collected digits match the extension of the SCCP IP phone, bypassing class of service configuration on both IP phones.
Answer : D
Explanation:
Cisco Type B SIP Phones offer functionality based SIP INVITE Message. Every key the end user presses triggers an individual SIP message. The first event is communicated with a SIP INVITE, but subsequent messages use SIP NOTIFY messages. The SIP NOTIFY messages send KPML events corresponding to any buttons or soft keys pressed by the user. Cisco Type B SIP IP Phones with SIP dial rules operate in the same manner as Cisco Type A phones with dial rules.
What does a comma accomplish when it is used in a SIP Dial Rule pattern that is associated with a Cisco 9971 IP Phone that is registered to Cisco Unified
Communications Manager?
- A. It inserts a 500-millisecond pause between digits.
- B. It causes the phone to generate a secondary dial tone.
- C. It is a delimiter and has no significant dialing impact.
- D. It indicates a timeout value of 5000 milliseconds.
- E. It is an obsolete parameter and will be ignored.
Answer : B
Explanation:
Comma is accepted in speed dial as delimiter and pause. -Comma used to delineate dial string, FAC, CMC, and post connect digits For post connect digits, commas insert a 2 second delay Commas may be duplicated to create longer delays.
What does a period accomplish when it is used in a SIP Dial Rule pattern that is associated with a Cisco 9971 IP Phone that is registered to Cisco Unified
Communications Manager?
- A. It matches any single digit from 0 to 9.
- B. It matches one or more digits from 0 to 9.
- C. It is a delimiter and has no significant dialing impact.
- D. It matches any single digit from 0 to 9, or the asterisk (*) or pound (#) symbols.
- E. It matches one or more digits from 0 to 9, or the asterisk (*) or pound (#) symbols.
Answer : D
Explanation:
Asterisk (*) matches one or more characters. The * gets processed as a wildcard character. You can override this by preceding the * with a backward slash (\) escape sequence, which results in the sequence \*. The phone automatically strips the \, so it does not appear in the outgoing dial string. When * is received as a dial digit, it gets matched by the wildcard characters * and period (.).
What does a weight represent in the Enhanced Location Call Admission Control mechanism on Cisco Unified Communications Manager?
- A. It defines the bandwidth that is available between locations.
- B. It defines the bandwidth that is available on a link.
- C. It is the amount of bandwidth allocation for different types of traffic.
- D. It is used to provide the relative priority of a link between locations.
- E. It is used to provide the relative priority of a location. D
Answer : Explanation
Explanation:
A weight provides the relative priority of a link in forming the effective path between any pair of locations. The effective path is the path used by Unified CM for the bandwidth calculations, and it has the least cumulative weight of all possible paths. Weights are used on links to provide a "cost" for the "effective path" and are pertinent only when there is more than one path between any two locations.
Which statement about the effective path in the Enhanced Location Call Admission Control mechanism on Cisco Unified Communications Manager is true?
- A. It is a sequence of links and intermediate locations that connect a pair of locations.
- B. It is used to define the bandwidth that is available between locations.
- C. Only one effective path is used between two locations.
- D. There could be multiple effective paths between a pair of locations.
- E. It logically represents the WAN link.
Answer : C
Explanation:
The effective path is the path used by Unified CM for the bandwidth calculations, and it has the least cumulative weight of all possible paths. Weights are used on links to provide a "cost" for the "effective path" and are pertinent only when there is more than one path between any two locations.
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