tag:blogger.com,1999:blog-17864550607766687782024-03-08T11:31:05.814-08:00CCNA Voice ccna huileCCNA Voice CCNA Voicejcreatorhttp://www.blogger.com/profile/17926347588908332596noreply@blogger.comBlogger60125tag:blogger.com,1999:blog-1786455060776668778.post-74425767712477359232015-07-16T17:30:00.001-07:002015-07-16T17:30:24.811-07:00Share your TUC Experience<div class="content" style="background-color: white; color: #333333; font-family: Verdana, 'BitStream vera Sans', Tahoma, Helvetica, sans-serif; font-size: 12px; line-height: 17.3999996185303px; margin: 0px; overflow: hidden; padding: 5px 0px 0px 5px;">
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As you know, the CCVP certification requires you to pass 5 exams: CVoice, QoS, CIPT1, CIPT 2 & TUC but currently this site has only materials for CCNA Voice IIUC 640-460 and CVoice 642-436. Many candidates have requested us to put up materials for other exams but it is a time-consuming work. In the mean time, we created the “Share your experience” for the TUC exam. We really hope anyone who read 9tut, digitaltut, certprepare, networktut and voicetut contribute to these sections as your experience is invaluable for CCVP learners to complete their goals.</div>
<div class="pinkandbold" style="color: magenta; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Please share with us your experience after taking the TUC exam, your materials, the way you learned, your recommendations…</div>
</div>
jcreatorhttp://www.blogger.com/profile/17926347588908332596noreply@blogger.com0tag:blogger.com,1999:blog-1786455060776668778.post-83902362034756118292015-07-16T17:29:00.003-07:002015-07-16T17:29:42.565-07:00Share your CIPT2 Experience<div class="content" style="background-color: white; color: #333333; font-family: Verdana, 'BitStream vera Sans', Tahoma, Helvetica, sans-serif; font-size: 12px; line-height: 17.3999996185303px; margin: 0px; overflow: hidden; padding: 5px 0px 0px 5px;">
<div style="margin-bottom: 10px; padding: 0px;">
As you know, the CCVP certification requires you to pass 5 exams: CVoice, QoS, CIPT1, CIPT 2 & TUC but currently this site has only materials for CCNA Voice IIUC 640-460 and CVoice 642-436. Many candidates have requested us to put up materials for other exams but it is a time-consuming work. In the mean time, we created the “Share your experience” for the CIPT2 exam. We really hope anyone who read 9tut, digitaltut, certprepare, networktut and voicetut contribute to these sections as your experience is invaluable for CCVP learners to complete their goals.</div>
<div class="pinkandbold" style="color: magenta; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Please share with us your experience after taking the CIPT2 exam, your materials, the way you learned, your recommendations</div>
</div>
jcreatorhttp://www.blogger.com/profile/17926347588908332596noreply@blogger.com0tag:blogger.com,1999:blog-1786455060776668778.post-46084547416302470632015-07-16T17:29:00.001-07:002015-07-16T17:29:02.454-07:00Share your CIPT1 Experience<div class="content" style="background-color: white; color: #333333; font-family: Verdana, 'BitStream vera Sans', Tahoma, Helvetica, sans-serif; font-size: 12px; line-height: 17.3999996185303px; margin: 0px; overflow: hidden; padding: 5px 0px 0px 5px;">
<div style="margin-bottom: 10px; padding: 0px;">
As you know, the CCVP certification requires you to pass 5 exams: CVoice, QoS, CIPT1, CIPT 2 & TUC but currently this site has only materials for CCNA Voice IIUC 640-460 and CVoice 642-436. Many candidates have requested us to put up materials for other exams but it is a time-consuming work. In the mean time, we created the “Share your experience” for the CIPT1 exam. We really hope anyone who read 9tut, digitaltut, certprepare, networktut and voicetut contribute to these sections as your experience is invaluable for CCVP learners to complete their goals.</div>
<div class="pinkandbold" style="color: magenta; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Please share with us your experience after taking the CIPT1 exam, your materials, the way you learned, your recommendations</div>
</div>
jcreatorhttp://www.blogger.com/profile/17926347588908332596noreply@blogger.com0tag:blogger.com,1999:blog-1786455060776668778.post-2264832919550185532015-07-16T17:27:00.001-07:002015-07-16T17:27:32.626-07:00Share your QoS Experience<div class="content" style="background-color: white; color: #333333; font-family: Verdana, 'BitStream vera Sans', Tahoma, Helvetica, sans-serif; font-size: 12px; line-height: 17.3999996185303px; margin: 0px; overflow: hidden; padding: 5px 0px 0px 5px;">
<div style="margin-bottom: 10px; padding: 0px;">
As you know, the CCVP certification requires you to pass 5 exams: CVoice, QoS, CIPT1, CIPT 2 & TUC but currently this site has only materials for CCNA Voice IIUC 640-460 and CVoice 642-436. Many candidates have requested us to put up materials for other exams but it is a time-consuming work. In the mean time, we created the “Share your experience” for the QoS exam. We really hope anyone who read 9tut, digitaltut, certprepare, networktut and voicetut contribute to these sections as your experience is invaluable for CCVP learners to complete their goals.</div>
<div class="pinkandbold" style="color: magenta; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Please share with us your experience after taking the QoS exam, your materials, the way you learned, your recommendations</div>
</div>
jcreatorhttp://www.blogger.com/profile/17926347588908332596noreply@blogger.com0tag:blogger.com,1999:blog-1786455060776668778.post-39118118878223601912015-07-16T17:26:00.003-07:002015-07-16T17:26:54.422-07:00Share your CAPPS v8.0 Experience<div class="content" style="background-color: white; color: #333333; font-family: Verdana, 'BitStream vera Sans', Tahoma, Helvetica, sans-serif; font-size: 12px; line-height: 17.3999996185303px; margin: 0px; overflow: hidden; padding: 5px 0px 0px 5px;">
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The new “Integrating Cisco Unified Communications Applications v8.0″ (CAPPS v8.0) 642-467 exam has come. This exam is new so it will need some time to put up materials for this exam. In the meantime, we create the “Share your CAPPS<br />v8.0 Experience” for everyone to share their experience after taking this exam.</div>
<div class="pinkandbold" style="color: magenta; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Please share with us your experience after taking the CAPPS v8.0 exam, your materials, the way you learned, your recommendations</div>
</div>
jcreatorhttp://www.blogger.com/profile/17926347588908332596noreply@blogger.com0tag:blogger.com,1999:blog-1786455060776668778.post-16949498738859904892015-07-16T17:26:00.001-07:002015-07-16T17:26:21.515-07:00Share your TVoice v8.0 Experience<div class="content" style="background-color: white; color: #333333; font-family: Verdana, 'BitStream vera Sans', Tahoma, Helvetica, sans-serif; font-size: 12px; line-height: 17.3999996185303px; margin: 0px; overflow: hidden; padding: 5px 0px 0px 5px;">
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The new “Troubleshooting Cisco Unified Communications v8.0″ (TVoice v8.0) 642-427 exam has come. This exam is new so it will need some time to put up materials for this exam. In the meantime, we create the “Share your TVoice v8.0 Experience” for everyone to share their experience after taking this exam.</div>
<div class="pinkandbold" style="color: magenta; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Please share with us your experience after taking the TVoice v8.0 exam, your materials, the way you learned, your recommendations</div>
</div>
jcreatorhttp://www.blogger.com/profile/17926347588908332596noreply@blogger.com0tag:blogger.com,1999:blog-1786455060776668778.post-557639994544482602015-07-16T17:25:00.004-07:002015-07-16T17:25:40.849-07:00Share your CIPT2 v8.0 Experience<div class="content" style="background-color: white; color: #333333; font-family: Verdana, 'BitStream vera Sans', Tahoma, Helvetica, sans-serif; font-size: 12px; line-height: 17.3999996185303px; margin: 0px; overflow: hidden; padding: 5px 0px 0px 5px;">
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The new CIPT2 v8.0 642-457 exam has come to replace for the CIPT2 642-456 exam. This exam is new so it will need some time to put up materials for this exam. In the meantime, we create the “Share your CIPT2 v8.0 Experience” for everyone to share their experience after taking this exam.</div>
<div class="pinkandbold" style="color: magenta; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Please share with us your experience after taking the CIPT2 v8.0 exam, your materials, the way you learned, your recommendations</div>
</div>
jcreatorhttp://www.blogger.com/profile/17926347588908332596noreply@blogger.com0tag:blogger.com,1999:blog-1786455060776668778.post-83334321584014754802015-07-16T17:25:00.000-07:002015-07-16T17:25:05.749-07:00Share your CIPT1 v8.0 Experience<div class="content" style="background-color: white; color: #333333; font-family: Verdana, 'BitStream vera Sans', Tahoma, Helvetica, sans-serif; font-size: 12px; line-height: 17.3999996185303px; margin: 0px; overflow: hidden; padding: 5px 0px 0px 5px;">
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The new CIPT1 v8.0 642-447 exam has come to replace for the CIPT1 642-446 exam. This exam is new so it will need some time to put up materials for this exam. In the meantime, we create the “Share your CIPT1 v8.0 Experience” for everyone to share their experience after taking this exam.</div>
<div class="pinkandbold" style="color: magenta; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Please share with us your experience after taking the CIPT1 v8.0 exam, your materials, the way you learned, your recommendations</div>
</div>
jcreatorhttp://www.blogger.com/profile/17926347588908332596noreply@blogger.com0tag:blogger.com,1999:blog-1786455060776668778.post-77667654863474884032015-07-16T17:23:00.002-07:002015-07-16T17:23:54.928-07:00Share your CVoice v8.0 Experience<div class="content" style="background-color: white; color: #333333; font-family: Verdana, 'BitStream vera Sans', Tahoma, Helvetica, sans-serif; font-size: 12px; line-height: 17.3999996185303px; margin: 0px; overflow: hidden; padding: 5px 0px 0px 5px;">
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The new “Implementing Cisco Unified Communications Voice over IP and QoS” v8.0 (CVoice v8.0) exam has come to replace the CVoice 642-436 and QoS 642-642 exams. This exam is new so it will need some time to put up materials for this exam. In the meantime, we create the “Share your CVoice v8.0 v8.0 Experience” for everyone to share their experience after taking this exam.</div>
<div class="pinkandbold" style="color: magenta; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Please share with us your experience after taking the CVoice v8.0 exam, your materials, the way you learned, your recommendations</div>
</div>
jcreatorhttp://www.blogger.com/profile/17926347588908332596noreply@blogger.com0tag:blogger.com,1999:blog-1786455060776668778.post-64682395986855926102015-07-16T17:19:00.005-07:002015-07-16T17:19:31.717-07:00Share your CVoice Experience<div class="content" style="background-color: white; color: #333333; font-family: Verdana, 'BitStream vera Sans', Tahoma, Helvetica, sans-serif; font-size: 12px; line-height: 17.3999996185303px; margin: 0px; overflow: hidden; padding: 5px 0px 0px 5px;">
<div class="pinkandbold" style="color: magenta; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Please share with us your experience after taking the CVoice 6.0 642-436 exam, your materials, the way you learned, your recommendations…</div>
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Your posts are warmly welcome!</div>
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Please don’t ask for links to download copyright materials here</div>
</div>
jcreatorhttp://www.blogger.com/profile/17926347588908332596noreply@blogger.com0tag:blogger.com,1999:blog-1786455060776668778.post-26472677523676150452015-07-16T17:19:00.002-07:002015-07-16T17:19:06.555-07:00Drag and Drop Questions<div class="content" style="background-color: white; color: #333333; font-family: Verdana, 'BitStream vera Sans', Tahoma, Helvetica, sans-serif; font-size: 12px; line-height: 17.3999996185303px; margin: 0px; overflow: hidden; padding: 5px 0px 0px 5px;">
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Here you will find answers to CVoice – Drag and Drop Questions</div>
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Question 1</div>
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The proper call-signaling term to the correct box in the diagram to establish RSVF-based Call Admission Control between the two Cisco Unifield Border Elements: Cisco UBEs. Some option is may be user more than once.</div>
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<img alt="RSVF-based Call Admission.jpg" border="0" height="450" src="http://voicetut.com/images/CVoice/DrapAndDrop/RSVF-based%20Call%20Admission.jpg" style="border: 0px; max-width: 800px;" width="600" /></div>
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Answer:</div>
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<img alt="RSVF-based_Call_Admission_Answer.jpg" height="338" src="http://voicetut.com/images/CVoice/DrapAndDrop/RSVF-based_Call_Admission_Answer.jpg" style="border: 0pt none; max-width: 800px;" width="600" /></div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
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Here is how the call is established with RSVF-based Call Admission Control</div>
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1) The Cisco Unified Communications Manager (at the left-side) sends an H.225 setup to the Cisco UBE.<br />2) The Cisco UBE processes the call setup information and associates an outbound VoIP dial peer requiring an RSVP reservation. The Cisco UBE sends out an RSVP Reservation request to the remote Cisco UBE.<br />3) The remote Cisco UBE acknowledges the reservation and initiates the reservation for the return path, which is acknowledged by the local Cisco UBE.<br />4) The H.225 setup message is routed to the remote Cisco UBE, which then routes the call to the outbound VoIP dial peer pointing to Cisco Unified Communications Manager (at the right-side).<br />5) H.245 negotiation occurs with media flow-through enabled.<br />6) The call is established.</div>
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Question 2</div>
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Click and drag the type of call on the above to the type of voice port it applies to on the below.</div>
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<img alt="voicePort.jpg" border="0" height="346" src="http://voicetut.com/images/CVoice/DrapAndDrop/voicePort.jpg" style="border: 0px; max-width: 800px;" width="500" /></div>
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Answer:</div>
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<img alt="voicePort_answer.jpg" border="0" height="222" src="http://voicetut.com/images/CVoice/DrapAndDrop/voicePort_answer.jpg" style="border: 0px; max-width: 800px;" width="495" /></div>
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1) T1 or E1 with CAS or PRI: PBX to PBX<br />2) FXO: off-net<br />3) FXS: local<br />4) FXS or switch: on-net<br />5) E&M, FXO, FXS: PLAR</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
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First let’s have a quick review of these types of calls:</div>
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<strong>Local calls</strong> are calls that occur when both the calling and called phones are attached to the same router.</div>
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<strong>On-net calls</strong> are calls that need more than one router. For example, the calling phone is from one router and the called phone attaches to another router. But notice that these routers are part of the same network.</div>
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<strong>Off-net calls</strong> are calls that originate on a router but terminate on the PSTN.</div>
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<strong>PLAR calls</strong> occur when a caller picks up a phone and the phone automatically dials a preconfigured number.</div>
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<strong>PBX to PBX calls</strong> are on-net calls, where the source and destination are PBXs.</div>
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Next we will explain the answers above:</div>
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<strong>PBX to PBX</strong> connections can use T1 or E1 with CAS or PRI: Nowadays, we often make PBX connections to a network through T1 or E1 lines with channel associated signaling (CAS) or Primary Rate Interface (PRI) signaling.</div>
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For<strong> off-net</strong> calls, the typical connection between the router and the PSTN is through FXO port.</div>
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A <strong>local</strong> call just need FXS ports so it is the only choice for this type of call.</div>
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We can make <strong>on-net</strong> calls through FXS port (phone directly connected to the router) or FXO port (phone connected to a PBX). The “switch” here means that we can connect an IP phone through a switch and place on-net calls through Cisco Unified Communications Manager.</div>
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A <strong>PLAR </strong>call can work with any type of signaling, including E&M, FXO, FXS interfaces.</div>
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Question 3</div>
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Assume a SIP voice network. Drag each characteristic to the type of SIP call setup the characteristics best describes.</div>
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<img alt="Type_of_SIP_call_setup.jpg" border="0" height="428" src="http://voicetut.com/images/CVoice/DrapAndDrop/Type_of_SIP_call_setup.jpg" style="border: 0px; max-width: 800px;" width="550" /></div>
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Answer:</div>
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<img alt="Type_of_SIP_call_setup_answer.jpg" border="0" height="250" src="http://voicetut.com/images/CVoice/DrapAndDrop/Type_of_SIP_call_setup_answer.jpg" style="border: 0px; max-width: 800px;" width="550" /></div>
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<strong>Direct call setup:</strong><br />+ Nonscalable<br />+ UA must keep data on large number of destinations<br />+ Relies on cached information to resolve addresses</div>
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<strong>Redirect Server Call Setup:</strong><br />+ Server reports back to a UA with destination coordinates</div>
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<strong>Proxy Server Call Setup:</strong><br />+ Most dynamic address resolution capability<br />+ All setup messages to through server<br />+ UA incapable of establishing its own sessions</div>
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Question 4</div>
<div style="margin-bottom: 10px; padding: 0px;">
Which item correctly describes the relationships between the feature and the category it belongs?</div>
<table border="1" cellpadding="2" style="background-attachment: initial; background-clip: initial; background-image: initial; background-origin: initial; background-position: initial; background-repeat: initial; background-size: initial; border-collapse: collapse; border: 2px solid rgb(204, 204, 204); margin: 5px 0px 10px;"><tbody>
<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">1</td><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">Supports analog faxes and modems on a VoIP network</td></tr>
<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">2</td><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">Performs call setup and teardown between VoIP networks and the PSTN</td></tr>
<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">3</td><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">Interconnects segments of the same or different VoIP networks using different media types</td></tr>
<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">4</td><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">Interconnects segments of the same or different VoIP networks using different signaling types</td></tr>
</tbody></table>
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A. Gateway – 1 and 2<br />CUBE – 3 and 4</div>
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B. Gateway – 1 and 3<br />CUBE – 2 and 4</div>
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C. Gateway – 2 and 3<br />CUBE – 1 and 4</div>
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D. Gateway – 2 and 4<br />CUBE – 1 and 3</div>
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(Note: In the real exam, this question may be represented as a drag and drop question)</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> A</div>
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jcreatorhttp://www.blogger.com/profile/17926347588908332596noreply@blogger.com0tag:blogger.com,1999:blog-1786455060776668778.post-27340427565897771772015-07-16T17:18:00.002-07:002015-07-16T17:18:35.844-07:00GateKeepers<div class="content" style="background-color: white; color: #333333; font-family: Verdana, 'BitStream vera Sans', Tahoma, Helvetica, sans-serif; font-size: 12px; line-height: 17.3999996185303px; margin: 0px; overflow: hidden; padding: 5px 0px 0px 5px;">
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Here you will find answers to CVoice – Gatekeeper Questions</div>
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Question 1</div>
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The SJ local zone contains a gatekeeper that controls two gateways, SJ1 and SJ2. Both gateways provide access to area code 408. Which two command strings should be entered into the gatekeeper to give the SJ2 gateway priority over the SJ1 gateway? (Choose two.)</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. zone prefix SJ 408 gw-priority 6 SJ1<br />B. zone prefix SJ 408 gw-priority 6 SJ2<br />C. zone prefix SJ 408 gw-priority 10 SJ1<br />D. zone prefix SJ 408 gw-priority 10 SJ2<br />E. zone prefix SJ 408 gw-priority 0 SJ2, 10 SJ1<br />F. zone prefix SJ 408 gw-priority 6 SJ1, 10 SJ2</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>A D</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
The simple syntax of “zone prefix” command is</div>
<div style="margin-bottom: 10px; padding: 0px;">
zone prefix <strong>gatekeeper-name</strong> <strong>e164-prefix</strong> [gw-priority priority gw-alias,…]</div>
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For example, the command “<strong>zone prefix SJ 408 gw-priority 6 SJ1</strong>“</div>
<div style="margin-bottom: 10px; padding: 0px;">
SJ is the gatekeeper-name, 408 is the E164-prefix area code, 6 is the priority and SJ1 is the GW-alias</div>
<div style="margin-bottom: 10px; padding: 0px;">
The [gw-priority priority gw-alias,…] part defines how the gatekeeper selects gateways in its local zone for calls to numbers beginning with prefix <strong>e164-prefix</strong>. The priority ranges from 0 to 10, where 0 prevents the gatekeeper from using the gateway gw-alias for that prefix and 10 places the highest priority on gateway gw-alias. The default is 5.</div>
<div style="margin-bottom: 10px; padding: 0px;">
By assigning SJ2 a priority value higher than that of SJ1, SJ2 will be the first choice when making call to this zone.</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 2</div>
<div style="margin-bottom: 10px; padding: 0px;">
Refer to the H.323 message in the exhibit. What is the gateway doing with the gatekeeper?</div>
<table align="center" border="0" style="background-attachment: initial; background-clip: initial; background-image: initial; background-origin: initial; background-position: initial; background-repeat: initial; background-size: initial; border-collapse: collapse; border: 2px solid rgb(204, 204, 204); margin: 5px 0px 10px;"><tbody>
<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">value RasMessage ::= registrationRequest :<div style="margin-bottom: 10px; padding: 0px;">
</div>
<div style="margin-bottom: 10px; padding: 0px;">
<br class="spacer_" /></div>
<div style="margin: 0px; padding: 0px;">
{<br />requestSeqNum 32633<br />protocolIdentifier { 0 0 8 2250 0 4}<br />discoveryComplete FALSE<br />callSignalAddress<br />{<br />}<br />rasAddress<br />{<div style="margin-bottom: 10px; padding: 0px;">
</div>
<div style="margin-bottom: 10px; padding: 0px;">
<br class="spacer_" /></div>
<div style="margin: 0px; padding: 0px;">
ipAddress :<br />{<br />ip ‘0A038201’H<br />port 53852<br />}</div>
<div style="margin-bottom: 10px; padding: 0px;">
}<br />terminalType<br />{</div>
<div style="margin: 0px; padding: 0px;">
mc FALSE<br />undefinedNode FALSE</div>
<div style="margin-bottom: 10px; padding: 0px;">
}<br />gatekeeperIdentifier {“HQ”}<br />endpointVendor<br />{</div>
<div style="margin: 0px; padding: 0px;">
vendor<br />{<div style="margin-bottom: 10px; padding: 0px;">
</div>
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<br class="spacer_" /></div>
<div style="margin: 0px; padding: 0px;">
t35countrycode 181<br />t35Extension 0<br />manufacturerCode 18</div>
<div style="margin-bottom: 10px; padding: 0px;">
}</div>
</div>
<div style="margin-bottom: 10px; padding: 0px;">
}<br />timeToLive 60<br />keepAlive TRUE<br />endpointIdentifier {“8452AEB4 00000002″}<br />willSupplyUUIEs FALSE<br />maintainConnection TRUE<br />}</div>
</div>
</td></tr>
</tbody></table>
<div style="margin-bottom: 10px; padding: 0px;">
A. initial registration<br />B. full registration<br />C. lightweight registration<br />D. registration retry</div>
<div style="margin-bottom: 10px; padding: 0px;">
<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>C</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
For the first time the gateway registers with the gatekeeper, it uses full registration. Prior to H.323 Version 2, Cisco gateways re-registered with the gatekeeper every 30 seconds. Each registration renewal used the same process as the initial registration, even though the gateway was already registered with the gatekeeper. This behavior generated considerable overhead at the gatekeeper. So from H.323 version 2, gateways can re-register with the gatekeeper using lightweight registration (it still requires the full registration process for initial registration, but uses an abbreviated renewal procedure to update the gatekeeper and minimize overhead).</div>
<div style="margin-bottom: 10px; padding: 0px;">
An endpoint’s registration with a gatekeeper may have a limited life span. The gatekeeper specifies the registration duration for an endpoint by including a timeToLive field in the Registration Confirm (RCF) message. After the specified length of time, the registration is considered expired. The endpoint must periodically send a Registration Request (RRQ) having the keepAlive bit set prior to the expiration time. Such a message may include a minimum amount of information as described in H.225.0 and is known as a lightweight RRQ.</div>
<div style="margin-bottom: 10px; padding: 0px;">
In the exhibit above, we can see the <strong>keepAlive</strong> bit is set to TRUE -> this is a lightweight RRQ.</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 3</div>
<div style="margin-bottom: 10px; padding: 0px;">
In which three RAS messages is the technology prefix sent? (Choose three.)</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. GRQ<br />B. RRQ<br />C. RCF<br />D. IRR<br />E. IRQ</div>
<div style="margin-bottom: 10px; padding: 0px;">
<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> A B E</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
The Cisco gatekeeper uses technology prefixes to group endpoints of the same type together. It uses the technology prefix appended in the called number to select the destination gateway or zone.</div>
<div style="margin-bottom: 10px; padding: 0px;">
This method prepends a technology prefix to the called number matched by the dial-peer. It is not used for registration, but for call setup with the Cisco gatekeeper. For example, called number 5551234 becomes 1#5551234.</div>
<div style="margin-bottom: 10px; padding: 0px;">
The technology prefix registration information is sent to the Cisco gatekeeper in the RAS Registration Request (RRQ) message. For example:</div>
<div style="margin-bottom: 10px; padding: 0px;">
GWY-B1(config)#interface ethernet 0/0<br />GWY-B1(config-if)#h323-gateway voip tech-prefix 1#</div>
<div style="margin-bottom: 10px; padding: 0px;">
-> B is correct.</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 4</div>
<div style="margin-bottom: 10px; padding: 0px;">
Refer to the output from the debug h225 asn1 command in the exhibit. You have configured a gatekeeper with two local zones, hq and br. You want the gateway at the branch location to register with zone BR. What needs to be corrected in the branch gateway to resolve the issue?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. Change the IP address in the h323-gateway voip id command.<br />B. Change the gatekeeper-id in the h323-gateway voip id command.<br />C. Add a zone remote for zone BR so the gateway can register with the correct zone.<br />D. Change the gatekeeper-id and the IP address in the h323-gateway voip id command.</div>
<div style="margin-bottom: 10px; padding: 0px;">
<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>B</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 5</div>
<div style="margin-bottom: 10px; padding: 0px;">
You have been asked to deploy a gatekeeper to support CUBE that will connect your organizational domain to the domain of an Internet Telephony Service Provider so that callers can reach the 407 area code. Which configuration will support this function?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A.<br />gatekeeper<br />zone local GKVIA acme.com 192.168.10.1<br />zone remote GK407 ITSP.com 10.10.1.100<br />zone prefix GK407…….<br />no shutdown</div>
<div style="margin-bottom: 10px; padding: 0px;">
B.<br />gatekeeper<br />zone local GKVIA acme.com 192.168.10.1<br />zone remote GK407 ITSP.com 10.10.1.100 1719 invia outvia GKVIA<br />zone prefix GK407 407*<br />no shutdown</div>
<div style="margin-bottom: 10px; padding: 0px;">
C.<br />gatekeeper<br />zone local GKVIA acme.com 192.168.10.1<br />zone remote GK407 ITSP.com 10.10.1.100 1719 invia GK407 outvia GK407<br />zone prefix GK407…….<br />no shutdown</div>
<div style="margin-bottom: 10px; padding: 0px;">
D.<br />gatekeeper<br />zone local GKVIA acme.com 192.168.10.1<br />zone remote GK407 ITSP.com 10.10.1.100 1719 invia GKVIA outvia GKVIA<br />zone prefix GK407 407*<br />no shutdown</div>
<div style="margin-bottom: 10px; padding: 0px;">
<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>D</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 6</div>
<div style="margin-bottom: 10px; padding: 0px;">
Refer to the exhibit. You have configured a gatekeeper and an IP-IP gateway on the same router. When you look at the output from the show gatekeeper endpoint command, the IP-IP gateway is not registered with the gatekeeper. What needs to be configured to resolve this issue?</div>
<div style="margin-bottom: 10px; padding: 0px; text-align: center;">
<img alt="sh_run.jpg" border="0" height="687" src="http://voicetut.com/images/CVoice/GateKeepers/sh_run.jpg" style="border: 0px; max-width: 800px;" width="388" /></div>
<div style="margin-bottom: 10px; padding: 0px; text-align: center;">
<img alt="show_gatekeeper_endpoint.jpg" border="0" height="459" src="http://voicetut.com/images/CVoice/GateKeepers/show_gatekeeper_endpoint.jpg" style="border: 0px; max-width: 800px;" width="492" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
A. You need to stop and restart the gateway.<br />B. You need to add a VoIP dial peer to the configuration.<br />C. The h323-gateway voip id command has an incorrect IP address.<br />D. The h323-gateway voip id command has an incorrect gatekeeper ID and IP address.</div>
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<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>B</div>
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Question 7</div>
<div style="margin-bottom: 10px; padding: 0px;">
Call Admission control (CAC) is a concept that applies to voice traffic only – not data traffic. Which two types are of Call Admission Control? (Choose two.)</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. resource-based<br />B. gatekeeper-controlled RSVP<br />C. local<br />D. QoS-based</div>
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<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>A C</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
There are 3 types of CAC:<br />+ Local CAC<br />+ Measurement Based CAC<br />+ Resource-Based CAC</div>
<div style="margin-bottom: 10px; padding: 0px;">
For more information about these types, please read: <a href="http://www.cisco.com/en/US/docs/ios/solutions_docs/voip_solutions/CAC.html" style="color: #2970a6; text-decoration: none;" target="_blank">http://www.cisco.com/en/US/docs/ios/solutions_docs/voip_solutions/CAC.html</a></div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 8</div>
<div style="margin-bottom: 10px; padding: 0px;">
The SJ local zone contains a gatekeeper that controls two gateways, SJ1 and SJ2. Both gateways provide access to area code 408. For the following command strings, which two will be entered into the gatekeeper to give the SJ2 gateway priority over the SJ1 gateway? (Choose two.)</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. zone prefix SJ 408 gw -priority 10 SJ2<br />B. zone prefix SJ 408 gw-priority 10 SJ1<br />C. zone prefix SJ 408 gw-priority 6 SJ2<br />D. zone prefix SJ 408 gw-priority 6 SJ1</div>
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<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>A D</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 9</div>
<div style="margin-bottom: 10px; padding: 0px;">
You are a Acme network administrator, your new task is to deploy a gatekeeper to support CUBE that will connect your organizational domain to the domain of an Internet Telephony Service Provider so that callers can reach the 407 area code. Which configuration will support this function?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A.<br />gatekeeper<br />zone local GKVIA acme.com 192.168.10.1<br />zone remote GK407 ITSP.com 10.10.1.100<br />zone prefix GK407 407<br />no shutdown</div>
<div style="margin-bottom: 10px; padding: 0px;">
B.<br />gatekeeper<br />zone local GKVIA acme.com 192.168.10.1<br />zone remote GK407 ITSP.com 10.10.1.100 1719 invia outvia GKVIA<br />zone prefix GK407 407*<br />no shutdown</div>
<div style="margin-bottom: 10px; padding: 0px;">
C.<br />gatekeeper<br />zone local GKVIA acme.com 192.168.10.1<br />zone remote GK407 ITSP.com 10.10.1.100 1719 invia GK407 outvia GK407<br />zone prefix GK407 407<br />no shutdown</div>
<div style="margin-bottom: 10px; padding: 0px;">
D.<br />gatekeeper<br />zone local GKVIA acme.com 192.168.10.1<br />zone remote GK407 ITSP.com 10.10.1.100 1719 invia GKVIA outvia GKVIA<br />zone prefix GK407 407*<br />no shutdown</div>
<div style="margin-bottom: 10px; padding: 0px;">
<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> D</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 10</div>
<div style="margin-bottom: 10px; padding: 0px;">
You are the director of the Acme VoIP network, based on the exhibit. You have a client that is testing a directory gatekeeper in the lab to provide address resolution between two different zones. Two of the commands in the running-config output are incorrect. Which two changes will correct the configuration? (Choose two.)</div>
<div style="margin-bottom: 10px; padding: 0px; text-align: center;">
<strong><img alt="directory_gatekeeper.jpg" border="0" height="468" src="http://voicetut.com/images/CVoice/GateKeepers/directory_gatekeeper.jpg" style="border: 0px; max-width: 800px;" width="588" /></strong></div>
<table border="0" style="background-attachment: initial; background-clip: initial; background-image: initial; background-origin: initial; background-position: initial; background-repeat: initial; background-size: initial; border-collapse: collapse; border: 2px solid rgb(204, 204, 204); margin: 5px 0px 10px;"><tbody>
<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;"><strong>voicetut# show running-config<br />…<br />!<br />zone local voicetut acm.com<br />zone local GK-A acme com 172.16.14.44 1719<br />zone remote GK-B acme.com 172.16.14.99 1719<br />zone prefix GK-A 770……<br />zone prefix GK-B 404….<br />no shutdown<br />!<br />…</strong></td></tr>
</tbody></table>
<div style="margin-bottom: 10px; padding: 0px;">
A.<br />replace<br />zone local GK-A acme.com 172.16.14.44 1719<br />with<br />zone remote GK-A acme.com 172.16.14.44 1719</div>
<div style="margin-bottom: 10px; padding: 0px;">
B.<br />replace<br />zone local DGK acme.com<br />with<br />zone remote DGK acme.com</div>
<div style="margin-bottom: 10px; padding: 0px;">
C.<br />replace<br />zone prefix GK-B 404….<br />with<br />zone prefix GK-B 404……..</div>
<div style="margin-bottom: 10px; padding: 0px;">
D.<br />replace<br />zone prefix GK-A 770…….<br />with<br />zone prefix GK-A 770….</div>
<div style="margin-bottom: 10px; padding: 0px;">
<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>A D</div>
</div>
jcreatorhttp://www.blogger.com/profile/17926347588908332596noreply@blogger.com0tag:blogger.com,1999:blog-1786455060776668778.post-79472709393770447512015-07-16T17:17:00.005-07:002015-07-16T17:17:53.819-07:00VoIP Design Elements<div class="content" style="background-color: white; color: #333333; font-family: Verdana, 'BitStream vera Sans', Tahoma, Helvetica, sans-serif; font-size: 12px; line-height: 17.3999996185303px; margin: 0px; overflow: hidden; padding: 5px 0px 0px 5px;">
<div style="margin-bottom: 10px; padding: 0px;">
Here you will find answers to CVoice – VoIP Design Element Questions</div>
<div style="margin-bottom: 10px; padding: 0px;">
</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 1</div>
<div style="margin-bottom: 10px; padding: 0px;">
Refer to the exhibit.</div>
<div style="margin-bottom: 10px; padding: 0px; text-align: center;">
<img alt="Calculate_number_of_calls.jpg" border="0" height="200" src="http://voicetut.com/images/CVoice/VoIPDesignElements/Calculate_number_of_calls.jpg" style="border: 0px; max-width: 800px;" width="550" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
Lighthorse Equine Management would like to investigate converging voice and data on their existing T1 Frame Relay WAN link between New York and Atlanta. Currently the following list of applications are consuming no more bandwidth than what is listed on this segment of the network.</div>
<div style="margin-bottom: 10px; padding: 0px;">
T1 link 1536 kbps<br />e-mail 75 kbps<br />internet 200 kbps<br />Oracle 500 kbps<br />FTP 250 kbps<br />Total 1025 kbps</div>
<div style="margin-bottom: 10px; padding: 0px;">
The customer has allocated 25% of the WAN link for routing updates and other overhead. They would like to increase the number of samples encapsulated in each PDU to 40 ms. You have calculated 6 bytes of overhead for Frame Relay, no cRTP, and the use of the G.711 codec. How many simultaneous calls could be placed on this link?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. 0 calls<br />B. 1 call<br />C. 2 calls<br />D. no more than 5 calls<br />E. no more than 10 calls<br />F. no more than 20 calls</div>
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<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>B</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 2</div>
<div style="margin-bottom: 10px; padding: 0px;">
Refer to the exhibit. A QoS strategy has already been deployed on the LAN. Choose three WAN QoS best practices that should be used over the WAN link. (Choose three.)</div>
<div style="margin-bottom: 10px; padding: 0px; text-align: center;">
<img alt="QoS.jpg" border="0" height="218" src="http://voicetut.com/images/CVoice/VoIPDesignElements/QoS.jpg" style="border: 0px; max-width: 800px;" width="625" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
A. Implement NBAR.<br />B. Implement admission control.<br />C. Mark voice traffic as EF in DSCP.<br />D. Mark voice traffic highest priority in 802.1p.<br />E. Use cRTP to maximize bandwidth utilization.<br />F. Configure access switches to trust traffic from IP phones.</div>
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<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>B C E</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 3</div>
<div style="margin-bottom: 10px; padding: 0px;">
Refer to the exhibit. Users are not able to complete a call from 678-555-1212 to 770-555-1111. What is the correct diagnosis for the problem?</div>
<div style="margin-bottom: 10px; padding: 0px; text-align: center;">
<img alt="dial-peer-4.jpg" border="0" height="600" src="http://voicetut.com/images/CVoice/DialPeers/dial-peer-4.jpg" style="border: 0px; max-width: 800px;" width="550" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
A. incorrect destination-pattern in router 1<br />B. incorrect POTS dial-peer statement in router 2<br />C. incorrect session-target statement in router 2<br />D. incorrect port statement in router 1 pots dial peer<br />E. missing no digit-strip on the voip dial peer in router 1</div>
<div style="margin-bottom: 10px; padding: 0px;">
<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> A</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 4</div>
<div style="margin-bottom: 10px; padding: 0px;">
A telemarketing firm needs to use number translation for incoming and outgoing calls. They have defined two translation profiles, one for incoming and one for outgoing calls. What can be used to simplify this task?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. dial peer<br />B. voice port<br />C. hunt group<br />D. trunk group<br />E. source IP group</div>
<div style="margin-bottom: 10px; padding: 0px;">
<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>D</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 5</div>
<div style="margin-bottom: 10px; padding: 0px;">
Which command parameter specifies that the router should not attempt to initiate a trunk connection but should wait for an incoming call before establishing the trunk?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. codec clear-channel<br />B. connection-trunk 404555…. answer-mode<br />C. voice-port 1/0:1<br />D. ds0-group timeslots 1-23 type ext-sig</div>
<div style="margin-bottom: 10px; padding: 0px;">
<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> B</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 6</div>
<div style="margin-bottom: 10px; padding: 0px;">
In a VoIP environment when speech samples are framed every 20 ms, a payload of 20 bytes is generated. Assuming a total packet length of 60 bytes, what is the length of the packet header if cRTP is deployed without redundancy checks?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. 1 byte<br />B. 2 bytes<br />C. 3 bytes<br />D. 4 bytes<br />E. 20 bytes<br />F. 40 bytes</div>
<div style="margin-bottom: 10px; padding: 0px;">
<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>B</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 7</div>
<div style="margin-bottom: 10px; padding: 0px;">
You have set up a complex dial plan using translation rules. The following translation rule has been configured. What output would correspond to the test translation-rule command?</div>
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<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;"><strong>translation-rule 1<br />rule 0 ^0.. 215550210<br />rule 1 ^1.. 215550211<br />rule 2 ^2.. 215550212<br />rule 3 ^3.. 215550213<br />rule 4 ^4.. 215550214<br />rule 5 ^5.. 215550215<br />rule 6 ^6.. 215550216<br />rule 7 ^7.. 215550217<br />rule 8 ^8.. 215550218<br />rule 9 ^9.. 215550210</strong></td></tr>
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A. test translation-rule 1 512<br />The replaced number: 21555021512<br />B. test translation-rule 1 555<br />The replaced number: 55521555021<br />C. test translation-rule 1 617<br />The replaced number: 61721555021<br />D. test translation-rule 1 910<br />The replaced number: 21555021910</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>A</div>
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Question 8</div>
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Which device is used to allow an H.323 stream to transit a firewall?</div>
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A. gatekeeper<br />B. gateway<br />C. proxy<br />D. MCU</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>C</div>
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Question 9</div>
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To hide its identity when initiating calls, Phone B requests that Server B place its calls for it. What kind of device is Server B?</div>
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<img alt="proxy.jpg" border="0" height="230" src="http://voicetut.com/images/CVoice/VoIPDesignElements/proxy.jpg" style="border: 0px; max-width: 800px;" width="600" /></div>
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A. proxy<br />B. redirect<br />C. registrar<br />D. user agent client<br />E. user agent server</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> A</div>
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jcreatorhttp://www.blogger.com/profile/17926347588908332596noreply@blogger.com0tag:blogger.com,1999:blog-1786455060776668778.post-77614520720569441972015-07-16T17:17:00.002-07:002015-07-16T17:17:15.602-07:00Dial Peers<div class="content" style="background-color: white; color: #333333; font-family: Verdana, 'BitStream vera Sans', Tahoma, Helvetica, sans-serif; font-size: 12px; line-height: 17.3999996185303px; margin: 0px; overflow: hidden; padding: 5px 0px 0px 5px;">
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Here you will find answers to CVoice – Dial Peer Questions</div>
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Question 1</div>
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Refer to the exhibit. You have been asked to configure a dial peer on R2 that will match only the extensions of the four telephones attached. Which dial-peer statement will you use?</div>
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<img alt="DialPeers1.jpg" border="0" height="350" src="http://voicetut.com/images/CVoice/DialPeers/DialPeers1.jpg" style="border: 0px; max-width: 800px;" width="600" /></div>
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A. dial-peer voice 1 pots<br />destination-pattern 5552.[0-5]0<br />B. dial-peer voice 1 pots<br />destination pattern 5552[5-6].0<br />C. dial-peer voice 1 pots<br />destination-pattern 555[2-5][56]<br />D. dial-peer voice 1 pots<br />destination-pattern 5552[5-6][05]0</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> D</div>
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Explanation</div>
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The numbers can be summaried as 5552(5 or 6)(5 or 0)0 so the destination-pattern should be written as 5552[5-6][05]0 or 5552[56][05]0</div>
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Question 2</div>
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Refer to the exhibit. When extension 201-555-1000 dials 404-555-1200, how are the digits manipulated in R1 so that they are presented correctly at R2?</div>
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<img alt="dial-peer-2.jpg" border="0" height="300" src="http://voicetut.com/images/CVoice/DialPeers/dial-peer-2.jpg" style="border: 0px; max-width: 800px;" width="550" /></div>
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<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;"><strong>hostname R1<br />!<br />interface serial 0/0<br />ip address 172.16.1.1 255.255.255.248<br />!<br />controller t1<br />framing esf<br />clock source line<br />lincode b8zs<br />ds0-group timeslots 1-24 type e&m-wink-start<br />!<br />voice-port 1/0:1<br />!<br />dial-peer voice 1 voip<br />destination-pattern 404555….<br />session-target ipv4:172.16.1.6<br />!<br />dial-peer voice 2 pots<br />destination-pattern 201555….<br />port 1/0:1</strong></td><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;"><strong>hostname R2<br />!<br />interface serial0/0<br />ip address 172.16.1.6 255.255.255.248<br />!<br />controller t1<br />framing esf<br />clock source line<br />lincode b8zs<br />ds0-group timeslots 1-24 type e&m-wink-start<br />!<br />voice-port 1/0:1<br />!<br />dial-peer voice 1 voip<br />destination-pattern 201555….<br />session-target ipv4:172.16.1.1<br />!<br />dial-peer voice 2 pots<br />destination-pattern 404555….<br />port 1/0:1</strong></td></tr>
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A. The outbound VoIP dial peer is matched and all digits are sent.<br />B. The digits 404-555 are stripped off before matching the outbound POTS dial peer.<br />C. The digits 404-555 are stripped off by the connection trunk and R2 receives only 1200.<br />D. R1 collects the 1200 and prepends the tie-line digits 404555. That number is matched to a VoIP dial peer and sent to the appropriate address.</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> A</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
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When 201-555-1000 (Phone A) calls 404-555-1200 (Phone B) the <strong>dial-peer voice 1 voip</strong> at R1 is matched with the destination-pattern 404555…. But notice that this is a <strong>voip</strong> dial-peer so digits are not stripped and all digits are sent to R2.</div>
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Question 3</div>
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Refer to the exhibit. Your customer wants to converge the existing PBX network with the IP network. The three remote offices have various types of PBXs. The customer is using a combination of tie-lines and trunks to connect the PBXs today. Which kind of connection should be implemented to allow calls to be placed from 201-555-1000 to 727-555-1000 so that when the call is completed, network resources are returned for other uses?</div>
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<img alt="dial-peer-3.jpg" border="0" height="400" src="http://voicetut.com/images/CVoice/DialPeers/dial-peer-3.jpg" style="border: 0px; max-width: 800px;" width="550" /></div>
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A. PLAR<br />B. trunk<br />C. tie-line<br />D. answer-mode</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>C</div>
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Explanation</div>
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E&M signaling supports tie-line type facilities.</div>
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Question 4</div>
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Which dial plan characteristic shows the most obvious improvement by dropping a number translation step?</div>
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A. availability<br />B. post-dial delay<br />C. scalability<br />D. hierarchical design</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>B</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
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Post-dial delay is the time between when the last digit is dialed and the moment the phone rings at the receiving location. In the PSTN, people expect a short post dial delay and to hear ring back within seconds. The more translations, digit manipulations, and lookups that take place, the longer the post dial delay becomes. Overall network design, translation rules, and alternate paths affect post dial delay. Minimize the amount of dial peers and translations to reduce post-dial delay.</div>
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By dropping a number translation step, the post-dial delay time will be obvious improvement.</div>
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Question 5</div>
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Refer to the exhibit. Users are not able to complete a call from 678-555-1212 to 770-555-1111. What is the correct diagnosis for the problem?</div>
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<img alt="dial-peer-4.jpg" border="0" height="600" src="http://voicetut.com/images/CVoice/DialPeers/dial-peer-4.jpg" style="border: 0px; max-width: 800px;" width="550" /></div>
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A. incorrect destination-pattern in router 1<br />B. incorrect POTS dial-peer statement in router 2<br />C. incorrect session-target statement in router 2<br />D. incorrect port statement in router 1 pots dial peer<br />E. missing no digit-strip on the voip dial peer in router 1</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>A</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
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The <strong>dial-peer 2 voip</strong> in Router 1 was configured “destination-pattern 770555..”. Notice that there are only two dots (.) in the destination-pattern that means when the user presses 770555<strong>11</strong>, the voip dial-peer is matched immediately without waiting for two last “11” pressed. Therefore router R2 only receives the “777055511” number and it doesn’t match with the destination-pattern in the “dial-peer voice 1 pots” configured in router R2.</div>
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Question 6</div>
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Refer to the exhibit. Three department managers share the directory number 3000. The Marketing manager’s phone is attached to port 1/1. The Engineering manager’s phone is attached to port 1/2. The Shipping manager’s phone is attached to port 1/3. In which situation would an incoming call ring on the Shipping manager’s phone?</div>
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<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;"><strong>dial-peer voice 1 pots</strong><div style="margin-bottom: 10px; padding: 0px;">
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<strong>destination pattern 3000<br />port 1/1<br />preference 0</strong></div>
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<strong>!<br />dial-peer voice 2 pots</strong></div>
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<strong>destination pattern 3000<br />port 1/2<br />preference 1</strong></div>
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<strong>!<br />dial-peer voice 3 pots</strong></div>
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<strong>destination pattern 3000<br />port 1/3<br />preference 2</strong></div>
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A. The Marketing manager is on the phone.<br />B. None of the managers are on the phone.<br />C. The Engineering manager is on the phone.<br />D. The Shipping manager and Marketing manager are on the phone.<br />E. The Engineering manager and Marketing manager are on the phone.</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> E</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
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With the <strong>preference 0</strong> configured in dial-peer voice 1 pots, this dial-peer (Marketing) has the highest priority to receive call if it is idle. Dial-peer 2 (Engineering) has the next priority and dial-peer 3 (Shipping) has lowest priority so it only rings when both Marketing and Engineering phones are busy.</div>
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It is a bit weird but the router considers lower preferences to be better than higher preferences. One more notice is that the default preference for a dial peer is 0.</div>
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Question 7</div>
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Refer to the exhibit. Your customers dial in to your company using a local number, and their calls cross the WAN to an IVR system. They are complaining that the IVR system does not always accept their input or may get it wrong. The IVR system has been checked and is working properly. What needs to be added to the dial peer on the incoming H.323 gateway to correct this problem?</div>
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<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;"><strong>dial-peer 100 voip</strong><div style="margin-bottom: 10px; padding: 0px;">
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<strong>destination-pattern …1111<br />session target ipv4:10.1.1.1<br />codec g729ar8</strong></div>
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A. no vad<br />B. tech-prefix 1#<br />C. codec g729ar8 bytes 30<br />D. dtmf-relay h245-alphanumeric</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>D</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
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DTMF is the tone generated when you press a button on a touch-tone phone. This tone is compressed at one end of a call; when the tone is decompressed at the other end, it can become distorted, depending on the codec used. The DTMF relay feature transports DTMF tones generated after call establishment out-of-band by using either a standard H.323 out-of-band method or a proprietary RTP-based mechanism. For session initiation protocol (SIP) calls, the most appropriate method to transport DTMF tones is Real-Time Transport Protocol named telephony event (RTP-NTE) or session initiation protocol notify (SIP Notify).</div>
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When you press a button on the touch-tone phone, a “high group” frequency is combined with a “low group” frequency and you can hear a generated tone. Notice that you often don’t see the “A B C D” column in most modern DTMF phones nowadays.</div>
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Although DTMF is usually transported accurately when using high-bit-rate voice codecs such as G.711, low-bit-rate codecs such as G.729 and G.723.1 are highly optimized for voice patterns and tend to distort DTMF tones. As a result, interactive voice response (IVR) systems may not correctly recognize the tones. Therefore the IVR sometimes can not recognize the DTMF tones and doesn’t accept their input o may get it wrong.</div>
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The main advantage of the “dtmf-relay” command is it sends DTMF tones with greater fidelity than is possible in-band for most low-bandwidth codecs, such as G.729 and G.723.</div>
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(Reference: CVoice Student Guide v6.0)</div>
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Question 8</div>
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You have designed a complex dial plan using digit manipulation. Given the following snippet of your configuration file, what action would you expect to result when a call beginning with the digits “612” is received?</div>
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<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;"><strong>dial-peer voice 1 pots<br />destination-pattern 612…..<br />no digit-strip<br />prefix 5501<br />port 1/0/0</strong></td></tr>
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A. A nine digit number beginning with 5501 will be forwarded.<br />B. A ten digit number beginning with 5501 will be forwarded.<br />C. A twelve digit number beginning with 5501612 will be forwarded.<br />D. A thirteen digit number beginning with 5501612 will be forwarded.</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>C</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
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This dial-peer has the “no digit-strip” command so no digits are stripped when this dial-peer is matched. So the whole number will be transferred with the format of <strong>5501612xxxxx</strong> (5501 is prefixed with the command <strong>prefix 5501</strong>)</div>
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Question 9</div>
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Which command sets parameters to search a series of dial peers for a destination that is not in use?</div>
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A. dial-peer rotary<br />B. dial-peer circulate<br />C. dial-peer hunt<br />D. dial-peer distribute</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>C</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
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Dial peer hunting is the process used when an originating router tries to establish a call on different dial peers if the originating router receives a user-busy invalid number or an unassigned-number disconnect cause code from a destination router.</div>
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Question 10</div>
<div style="margin-bottom: 10px; padding: 0px;">
On the basis of the provided exhibit. Enzo’s Bikes manufactures high end bicycle frames. Until recently they sold only to bicycle shops; however, now they are starting to sell to end users. They need a way to add two additional sales staff and ensure that the senior sales technician always gets the first call. Drew is the senior sales technician. Bob is the newest sales technician. Bob’s phone should always be the last one chosen for incoming sales calls, after Drew and James. Bob’s phone should be chosen first only when Drew and James are busy on calls. Select the correct dial-peer command set for Bob’s phone.</div>
<div style="margin-bottom: 10px; padding: 0px; text-align: center;">
<img alt="dial_peer_preference.jpg" border="0" height="450" src="http://voicetut.com/images/CVoice/DialPeers/dial_peer_preference.jpg" style="border: 0px; max-width: 800px;" width="650" /></div>
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A.<br />dial-peer voice 3 pots<br />destination-pattern 5555110<br />preference 2</div>
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B.<br />dial-peer voice 3 pots<br />destination-pattern 5555110<br />preference firstlast</div>
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C.<br />dial-peer voice 3 pots<br />destination-pattern 5555110<br />preference 0</div>
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D.<br />dial-peer voice 3 pots<br />destination-pattern 5555110<br />preference high</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>A</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
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The router considers lower preferences to be better than higher preferences and the default preference is 0. Therefore, by setting the preference of Bob’s dial-peer to 2 we guarantee Bob will be the last one to receive the call (while James’ priority is set to 1 and Drew uses the default configuration).</div>
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jcreatorhttp://www.blogger.com/profile/17926347588908332596noreply@blogger.com0tag:blogger.com,1999:blog-1786455060776668778.post-80005041738104460392015-07-16T17:16:00.003-07:002015-07-16T17:16:32.295-07:00Advanced Dial Plans<div class="content" style="background-color: white; color: #333333; font-family: Verdana, 'BitStream vera Sans', Tahoma, Helvetica, sans-serif; font-size: 12px; line-height: 17.3999996185303px; margin: 0px; overflow: hidden; padding: 5px 0px 0px 5px;">
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Here you will find answers to CVoice – Advanced Dial Plan Questions</div>
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Question 1</div>
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Which mechanism do you use to implement calling privileges on Cisco Unified Communications Manager Express?</div>
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A. CoS<br />B. QoS<br />C. CAC<br />D. COR<br />E. SRST</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> D</div>
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Explanation</div>
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Calling privileges define the destination a user is allowed to dial and they are implemented on Cisco IOS gateway using Class of Restriction.</div>
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Class of Restriction (COR) is the feature that determines which numbers might not be dialed on the system. COR is required only when you want to restrict the ability of some phones to make certain types of calls but allow other phones to place those calls. COR functionality provides the ability to deny certain call attempts on the basis of the incoming and outgoing CORs that are provisioned on the dial-peers. This functionality provides flexibility in network design, allows users to block calls (for example, calls to 900 numbers), and applies different restrictions to call attempts from different originators.</div>
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(Reference: <a href="http://www.cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a008019d649.shtml" rel="nofollow" style="color: #2970a6; text-decoration: none;" target="_blank">http://www.cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a008019d649.shtml</a>)</div>
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Question 2</div>
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Using Cisco Unified Communications Manager Express, what four steps are necessary to implement COR? (Choose four)</div>
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A. Configure SRST.<br />B. Define COR labels.<br />C. Configure COR lists.<br />D. Assign COR list to ephone-DN.<br />E. Configure COR lists on voice ports.<br />F. Configure dial peers and assign COR lists.</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> B C D F</div>
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Explanation</div>
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Four steps to configure COR on Cisco IOS gateway using Cisco Unified Communications Manager Express:</div>
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1) Define COR labels.<br />2) Configure COR lists.<br />3) Configure dial peers and assign COR lists.<br />4) Assign COR lists to ephone-dn.</div>
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For example, we will define three calling privilege classes:</div>
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<strong>Local:</strong> This class should allow emergency and local calls.<br /><strong>Long Distance:</strong> This class should allow emergency, local, and long distance calls.<br /><strong>International:</strong> This class should allow emergency, local, long distance, and international calls.</div>
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Step 1: Define the four COR labels to be used as COR list members with the command <strong>dial-peer cor custom</strong>.</div>
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Router(config)#dial-peer cor custom<br />Router(config-dp-cor)#name 911<br />Router(config-dp-cor)#name local<br />Router(config-dp-cor)#name ld<br />Router(config-dp-cor)#name intl</div>
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<strong>Description</strong></div>
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<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">911: Allows calls to emergency 911</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">local: Allows local calls only</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">ld: Allows long distance calls</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">intl: Allows international calls</li>
</ul>
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Step 2: Define the COR lists that will be assigned as “outgoing” to the PSTN dial peers with the command <strong>dial-peer cor list</strong>.</div>
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Router(config-dp-corlist)#dial-peer cor list 911call<br />Router(config-dp-corlist)#member 911<br />Router(config-dp-corlist)#dial-peer cor list localcall<br />Router(config-dp-corlist)#member local<br />Router(config-dp-corlist)#dial-peer cor list ldcall<br />Router(config-dp-corlist)#member ld<br />Router(config-dp-corlist)#dial-peer cor list intlcall<br />Router(config-dp-corlist)#member intl</div>
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Define the COR lists that will be assigned as “incoming” from the local dial peers with the command <strong>dial-peer cor list </strong>.</div>
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Router(config)#dial-peer cor list local<br />Router(config-dp-corlist)#member 911<br />Router(config-dp-corlist)#member local</div>
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Router(config)#dial-peer cor list ld<br />Router(config-dp-corlist)#member 911<br />Router(config-dp-corlist)#member local<br />Router(config-dp-corlist)#member ld</div>
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Router(config)#dial-peer cor list intl<br />Router(config-dp-corlist)#member 911<br />Router(config-dp-corlist)#member local<br />Router(config-dp-corlist)#member ld<br />Router(config-dp-corlist)#member intl</div>
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Step 4: Assign Outbound COR Lists to PSTN Dial Peers</div>
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<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">Dial peer 911 has the outgoing 911call COR list</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">Dial peer 9911 has the outgoing 911call COR list.</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">Dial peer 9 has the outgoing localcall COR list.</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">Dial peer 91 has the outgoing ldcall COR list.</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">Dial peer 9011 has the outgoing intlcall COR list.</li>
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Router(config)#dial-peer voice 911 pots<br />Router(config-dial-peer)#destination-pattern 911<br />Router(config-dial-peer)#forward-digits all<br />Router(config-dial-peer)#corlist outgoing 911call<br />Router(config-dial-peer)#port 0/0/0:23</div>
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Router(config)#dial-peer voice 9911 pots<br />Router(config-dial-peer)#destination-pattern 9911<br />Router(config-dial-peer)#forward-digits 3<br />Router(config-dial-peer)#corlist outgoing 911call<br />Router(config-dial-peer)#port 0/0/0:23</div>
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Router(config)#dial-peer voice 9 pots<br />Router(config-dial-peer)#destination-pattern 9[2-9]……<br />Router(config-dial-peer)#corlist outgoing localcall<br />Router(config-dial-peer)#port 0/0/0:23</div>
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Router(config)#dial-peer voice 91 pots<br />Router(config-dial-peer)#destination-pattern 91[2-9]..[2-9]……<br />Router(config-dial-peer)#prefix 1<br />Router(config-dial-peer)#corlist outgoing ldcall<br />Router(config-dial-peer)#port 0/0/0:23</div>
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Router(config)#dial-peer voice 9011 pots<br />Router(config-dial-peer)#destination-pattern 9011T<br />Router(config-dial-peer)#prefix 011<br />Router(config-dial-peer)#corlist outgoing intlcall<br />Router(config-dial-peer)#port 0/0/0:23</div>
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<em>Reference: CVoice Student Guide v6.0 (Page 4-165)</em></div>
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Question 3</div>
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Refer to the exhibit. Which dial peer configuration will block phone A from making long distance calls?</div>
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<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;"><strong>Partial configuration on Gateway-A:</strong><div style="margin-bottom: 10px; padding: 0px;">
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dial-peer cor custom</div>
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name Emergency<br />name Local<br />name LD<br />name Intl</div>
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dial-peer cor list Em01</div>
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member Emergency</div>
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dial-peer cor list Local01</div>
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member Local</div>
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dial-peer cor list LD01</div>
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member LD</div>
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dial-peer cor list Intl01</div>
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member Intl</div>
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dial-peer cor list LocalLst</div>
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member Emergency<br />member Local</div>
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dial-peer cor list LDLst</div>
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member Emergency<br />member Local<br />member LD</div>
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dial-peer cor list IntlLst</div>
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member Emergency<br />member Local<br />member LD<br />member Intl</div>
</td><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;"><img alt="cor_lists.jpg" border="0" height="260" src="http://voicetut.com/images/CVoice/AdvancedDialPlans/cor_lists.jpg" style="border: 0px; max-width: 800px;" width="600" /></td></tr>
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A.<br />dial-peer voice 1374 pots<br />destination-pattern 1374<br />port1/0/0<br />dial-peer voice 100 voip<br />corlist incoming Intl01<br />destination-pattern 9011T<br />session target ipv4:192.168.101.254</div>
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B.<br />dial-peer voice 1374 pots<br />destination-pattern 1374<br />port1/0/0<br />dial-peer voice 100 voip<br />corlist outgoing Intl01<br />destination-pattern 9011T<br />session target ipv4:192.168.101.254</div>
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C.<br />dial-peer voice 1374 pots<br />corlist incoming LDLst<br />destination-pattern 1374<br />port1/0/0<br />dial-peer voice 100 voip<br />destination-pattern 9011T<br />session target ipv4:192.168.101.254</div>
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D.<br />dial-peer voice 1374 pots<br />corlist outgoing LDLst<br />destination-pattern 1374<br />port1/0/0<br />dial-peer voice 100 voip<br />destination-pattern 9011T<br />session target ipv4:192.168.101.254</div>
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E.<br />dial-peer voice 1374 pots<br />corlist incoming LocalLst<br />destination-pattern 1374<br />port1/0/0<br />dial-peer voice 100 voip<br />corlist outgoing Intl01<br />destination-pattern 9011T<br />session target ipv4:192.168.101.254</div>
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F.<br />dial-peer voice 1374 pots<br />corlist outgoing LDLst<br />destination-pattern 1374<br />port1/0/0<br />dial-peer voice 100 voip<br />corlist outgoing Intl01<br />destination-pattern 9011T<br />session target ipv4:192.168.101.254</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> E</div>
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Explanation</div>
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To block phone A from making long distance calls, phone A must belong to an “incoming” dial-peer which is not a member of the LD (Long Distance). In three incoming dial-peer (the three last dial-peers), there is only one dial-peer satisfies with this condition, that is the <strong>LocalLst </strong>dial-peer so the answer should be E.</div>
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One tip to quickly recognizes which dial-peer is for “outgoing” dial-peer is that this type of dial-peer usually have only one member. In this question, the outgoing dial-peers are Em01, Local01, LD01, Intl01. Dial-peers which have more than one member are often “incoming” dial-peers.</div>
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You can read <a href="http://www.cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a008019d649.shtml" rel="nofollow" style="color: #2970a6; text-decoration: none;" target="_blank">http://www.cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a008019d649.shtml</a> for another example. [/protect]</div>
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Question 4</div>
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Where would you assign COR lists in Cisco Unified Communications Manager Express?</div>
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A. ephone<br />B. ephone-dn<br />C. voice register dn<br />D. voice register pool</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> B</div>
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Explanation</div>
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For Cisco Unified Communications Manager Express, the COR list is directly assigned to the appropriate Ethernet phone-dn (ephone-directory number)</div>
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jcreatorhttp://www.blogger.com/profile/17926347588908332596noreply@blogger.com0tag:blogger.com,1999:blog-1786455060776668778.post-1954416767817280552015-07-16T17:15:00.003-07:002015-07-16T17:15:35.779-07:00Digital Voice Ports<div class="content" style="background-color: white; color: #333333; font-family: Verdana, 'BitStream vera Sans', Tahoma, Helvetica, sans-serif; font-size: 12px; line-height: 17.3999996185303px; margin: 0px; overflow: hidden; padding: 5px 0px 0px 5px;">
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Here you will find answers to CVoice – Digital Voice Port Questions</div>
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Question 1</div>
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A customer needs to configure a CAS E&M circuit that will support inbound and outbound DNIS and inbound ANI. Which configuration will accomplish this task?</div>
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A. pri-group timeslots 1-24<br />B. ds0-group 0 timeslots 1-24 type none<br />C. ds0-group 0 timeslots 1-24 type e&m-fgd<br />D. ds0-group 0 timeslots 1-24 type fgd-eana<br />E. ds0-group 0 timeslots 1-31 type r2-digital r2-compelled ani</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>C</div>
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Explanation</div>
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To define T1 channels for compressed voice calls and the channel-associated signaling (CAS) method by which the router connects to the PBX or PSTN, enter the <strong>ds0-group</strong> controller configuration command. Below is the syntax of this command:</div>
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<strong>ds0-group</strong> <em>ds0-group-no</em> <strong>timeslots </strong><em>timeslot-list</em> <strong>type </strong>{<strong>e&m-immediate</strong> | <strong>e&m-delay</strong> | <strong>e&m-wink</strong> | <strong>e&m-fgd</strong> |<strong> fgd-eana</strong>}</div>
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<strong>Description</strong></div>
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<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">ds0-group-no</td><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">A value from 0 to 23 that identifies the DS0 group</td></tr>
<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">timeslot-list</td><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">timeslot-list is a single timeslot number, a single range of numbers, or multiple ranges of numbers separated by commas. For T1, allowable values are from 1 to 24. Examples are:<div style="margin-bottom: 10px; padding: 0px;">
</div>
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<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">2</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">1-15, 17-24</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">1-23</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">2, 4, 6-12</li>
</ul>
</td></tr>
<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">type</td><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">The signaling method selection for <strong>type </strong>depends on the connection that you are making. The E&M interface allows connection for PBX trunk lines (tie lines) and telephone equipment.<div style="margin-bottom: 10px; padding: 0px;">
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The options are as follows:</div>
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<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;"><strong>e&m-immediate-start </strong>specifies no specific offhook and onhook signaling.</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;"><strong>e&m-delay </strong>specifies that the originating endpoint sends an offhook signal and then and waits for an offhook signal followed by an onhook signal from the destination.</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;"><strong>e&m-fgd</strong> specifies E&M Type II Feature Group D.</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;"><strong>e&m-wink-start </strong>specifies that the originating endpoint sends an offhook signal and waits for a wink signal from the destination.</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;"><strong>fgd-eana</strong> specifies Group D Exchange Access North American (EANA) signaling.</li>
</ul>
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(There are some more options but they are omitted)</div>
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(Reference: <a href="http://www.ciscosecure.net/en/US/docs/ios/12_1/12_1xd/feature/guide/hdv_fgd.html" rel="nofollow" style="color: #2970a6; text-decoration: none;" target="_blank">http://www.ciscosecure.net/en/US/docs/ios/12_1/12_1xd/feature/guide/hdv_fgd.html</a>)</div>
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T1 CAS always provides the ANI/DNIS delimiter on incoming T1/CAS trunk lines. The customer wants E&M circuit so the answer should be C.</div>
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<strong>Notice:</strong></div>
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+ CAS signaling main feature is its use of user bandwidth to perform signaling functions. CAS signaling is often referred to as robbed-bit-signaling because user bandwidth is being “robbed” by the network for other purposes.</div>
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+<strong> E&M signaling</strong> is typically used for trunks. It is normally the only way that a central office (CO) switch can provide two-way dialing with direct inward dialing.</div>
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+ <strong>ANI</strong> – Automatic number identification. SS7 (signaling system 7) feature in which a series of digits, either analog or digital, are included in the call, identifying the telephone number of the calling device. In other words, ANI identifies the number of the calling party.</div>
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+ <strong>DNIS</strong> – Dialed number identification service, also known as the called party number. The telephone number of the called party after translation occurs in the Public Switched Telephone Network. A given destination may have a different DNIS number based on how the call is placed (for example, 800 or direct dial).</div>
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Question 2</div>
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In T1 CAS, where are the signaling states and control features carried for Super Frame robbed-bit signaling?</div>
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A. 6th and 12th frame<br />B. 6th, 12th, 18th, and 24th frame<br />C. the first and seventeenth time slot<br />D. the first and sixteenth time slot</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>A</div>
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Explanation</div>
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Each T1 has 24 channels ( or 24 DS0 – digital signal level 0) that can transmit 8 bits per channel each. This give us a total of 192 bits. One more bit is used for framing, bringing the total to 193 bits. Super Frame bundles 12 of these 193-bit frames for transport. The picture below shows the structure of a T1 Super Frame</div>
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<img alt="T1_CAS_Super_Frame.jpg" src="http://www.voicetut.com/images/CVoice/DigitalVoicePorts/T1_CAS_Super_Frame.jpg" style="border: 0pt none; max-width: 800px;" /></div>
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The T1 CAS signaling looks at every 6th & 12th frames for signaling information, these bits are referred to as the A and B bits. The A and B bits can represent different signaling states or control features (on hook or off hook, idle, busy, ringing, and addressing)</div>
<div style="margin-bottom: 10px; padding: 0px; text-align: center;">
<img alt="T1_CAS_Super_Frame_6th_12th.jpg" height="300" src="http://www.voicetut.com/images/CVoice/DigitalVoicePorts/T1_CAS_Super_Frame_6th_12th.jpg" style="border: 0pt none; max-width: 800px;" width="550" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
According to Nyquist theory, we sample voice 8000 times per second, that means we need to send 8000 of these 193-bit frames every second. So 8000 x 193 = 1,544 Mbps.</div>
<div style="margin-bottom: 10px; padding: 0px;">
Extended super frame (ESF), due to grouping the frames in sets of twenty-four, has four signaling bits per channel or timeslot. These occur in frames 6, 12, 18, and 24 and are called the A-, B-, C-, and D-bits respectively. So if the question asks about ESF, the answer should be B.</div>
</div>
jcreatorhttp://www.blogger.com/profile/17926347588908332596noreply@blogger.com0tag:blogger.com,1999:blog-1786455060776668778.post-85452084012985534532015-07-16T17:14:00.002-07:002015-07-16T17:14:26.685-07:00VoIP Gateways<div class="content" style="background-color: white; color: #333333; font-family: Verdana, 'BitStream vera Sans', Tahoma, Helvetica, sans-serif; font-size: 12px; line-height: 17.3999996185303px; margin: 0px; overflow: hidden; padding: 5px 0px 0px 5px;">
<div style="margin-bottom: 10px; padding: 0px;">
Here you will find answers to CVoice – VoIP Gateway Questions</div>
<div style="margin-bottom: 10px; padding: 0px;">
</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 1</div>
<div style="margin-bottom: 10px; padding: 0px;">
Refer to the exhibit. Choose the correct configuration command set that will allow the gateway in zone BR to register with the gatekeeper in the same zone.</div>
<div style="margin-bottom: 10px; padding: 0px; text-align: center;">
<img alt="Gateway_register_gatekeeper.jpg" border="0" height="278" src="http://voicetut.com/images/CVoice/VoIPGateways/Gateway_register_gatekeeper.jpg" style="border: 0px; max-width: 800px;" width="482" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
A.<br />interface fastethernet 0/0<br />ip address 10.1.110.1<br />h323-gateway voip interface<br />h323-gateway voip id BR ipaddr 10.2.120.1<br />h323-gateway voip h323-id BRgw<br />!<br />gateway<br />B.<br />interface fastethernet 0/0<br />ip address 10.2.120.1<br />h323-gateway voip interface<br />h323-gateway voip id BR ipaddr 10.1.120.2<br />h323-gateway voip h323-id BRgw<br />!<br />gateway<br />C.<br />interface fastethernet 0/0<br />ip address 10.2.120.1<br />h323-gateway voip interface<br />h323-gateway voip id BR ipaddr 10.2.120.1<br />h323-gateway voip h323-id BRgw<br />!<br />gateway<br />D.<br />interface fastethernet 0/0<br />ip address 10.1.110.1<br />h323-gateway voip interface<br />h323-gateway voip id BR ipaddr 10.1.110.1<br />h323-gateway voip h323-id BRgw<br />!<br />gateway<br />E.<br />interface fastethernet 0/0<br />ip address 10.2.120.1<br />h323-gateway voip interface<br />h323-gateway voip id HQ ipaddr 10.1.110.1<br />h323-gateway voip h323-id BRgw<br />!<br />gateway</div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>C</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
Notice that the router at zone Br is functioned as both gateway and gatekeeper and it uses the IP address of 10.2.120.1 as the “zone local BR”. Therefore if we want “the gateway in zone BR to register with the gatekeeper in the same zone” we must use 10.2.120.1 in the command:</div>
<div style="margin-bottom: 10px; padding: 0px;">
<strong>h323-gateway voip id BR ipaddr 10.2.120.1</strong></div>
<div style="margin-bottom: 10px; padding: 0px;">
In which, BR <strong></strong>is the zone name defined in the “zone local BR” command (of the gatekeeper) and the <strong></strong>is the IP address of an interface of the gatekeeper and it should be 10.2.120.1-> B, D and E are not correct.</div>
<div style="margin-bottom: 10px; padding: 0px;">
A can be correct but it is not as clear as answer C.</div>
<div style="margin-bottom: 10px; padding: 0px;">
Notice that last command <strong>h323-gateway voip h323-id BRgw</strong> specifies the <strong>BRgw </strong>is the name of the gateway to communicate with the gatekeeper.</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 2</div>
<div style="margin-bottom: 10px; padding: 0px;">
Examine the example output.</div>
<table border="0" style="background-attachment: initial; background-clip: initial; background-image: initial; background-origin: initial; background-position: initial; background-repeat: initial; background-size: initial; border-collapse: collapse; border: 2px solid rgb(204, 204, 204); margin: 5px 0px 10px;"><tbody>
<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;"><strong>hostname GW1<br />!<br />interface Ethernet 0/0<br />ip address 172.16.2.1 255.255.255.0<br />h323-gateway voip interface<br />h323-gateway voip id GK1-zone1.abc.com abc.com ipaddr 172.16.2.2<br />h323-gateway voip h323-id GW1<br />h323-gateway voip bind srcaddr 172.16.2.1<br />!<br />dial-peer voice 1 voip<br />destination-pattern 1212…….<br />session-target ras<br />!<br />dial-peer voice 2 pots<br />destination-pattern 2125551212<br />no register e164<br />!<br />end</strong></td></tr>
</tbody></table>
<div style="margin-bottom: 10px; padding: 0px;">
Choose the command that will restore communication with gatekeeper functionality to this device.</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. h323-gateway voip h323-id GK1<br />B. gateway<br />C. h323-gateway voip bind srcaddr 172.16.2.2<br />D. h323-gateway voip GW1-zone2.abc.com abc.com ipaddr 172.16.2.1</div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> B</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
The <strong>gateway </strong>command enables the H.323 VoIP gateway to register with the gatekeeper. This is the first command you should enter when configuring a voice gateway.</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 3</div>
<div style="margin-bottom: 10px; padding: 0px;">
Which item correctly describes the relationships between the feature and the category it belongs?</div>
<table border="1" cellspacing="3" style="background-attachment: initial; background-clip: initial; background-image: initial; background-origin: initial; background-position: initial; background-repeat: initial; background-size: initial; border-collapse: collapse; border: 2px solid rgb(204, 204, 204); margin: 5px 0px 10px;"><tbody>
<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">1</td><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">Supports analog faxes and modems on a VoIP network</td></tr>
<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">2</td><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">Performs call setup and teardown between VoIP networks and the PSTN</td></tr>
<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">3</td><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">Interconnects segments of the same or different VoIP networks using different media types</td></tr>
<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">4</td><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">Interconnects segments of the same or different VoIP network using different signaling types</td></tr>
</tbody></table>
<div style="margin-bottom: 10px; padding: 0px;">
A. Gateway – 1 and 2<br />CUBE – 3 and 4<br />B. Gateway – 1 and 3<br />CUBE – 2 and 4<br />C. Gateway – 2 and 3<br />CUBE – 1 and 4<br />D. Gateway – 2 and 4<br />CUBE – 1 and 3</div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>A</div>
</div>
jcreatorhttp://www.blogger.com/profile/17926347588908332596noreply@blogger.com0tag:blogger.com,1999:blog-1786455060776668778.post-16773543403990257822015-07-16T17:13:00.002-07:002015-07-16T17:13:56.687-07:00Call Routing and Path Selection<div class="content" style="background-color: white; color: #333333; font-family: Verdana, 'BitStream vera Sans', Tahoma, Helvetica, sans-serif; font-size: 12px; line-height: 17.3999996185303px; margin: 0px; overflow: hidden; padding: 5px 0px 0px 5px;">
<div style="margin-bottom: 10px; padding: 0px;">
Here you will find answers to CVoice – Call Routing and Path Selection Questions</div>
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</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 1</div>
<div style="margin-bottom: 10px; padding: 0px;">
When setting up a VoIP call, what is the first thing a gateway router tries to match to a dialed number?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. call leg<br />B. IP route<br />C. session target<br />D. destination pattern</div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>D</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
First, the gateway attempts to match the called number with the<strong> incoming called-number</strong>. If no match is found, the router or gateway attempts to match the calling number of the call set-up request with the <strong>answer-address</strong> of each dial-peers. If no match is found, it attempts to match the calling number of the call set-up request to the <strong>destination-pattern</strong> of each dial-peer.</div>
<div style="margin-bottom: 10px; padding: 0px;">
Notice that these steps are just applied for inbound dial peer.</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 2</div>
<div style="margin-bottom: 10px; padding: 0px;">
Refer to the exhibit. Highland Park Property Development is integrating a Cisco Unified Communications Manager Express system with the existing PBX via an E1 QSIG trunk. After the initial configuration, no calls can be placed from IP phones to PBX phones. How can this problem be resolved?</div>
<table border="0" style="background-attachment: initial; background-clip: initial; background-image: initial; background-origin: initial; background-position: initial; background-repeat: initial; background-size: initial; border-collapse: collapse; border: 2px solid rgb(204, 204, 204); margin: 5px 0px 10px;"><tbody>
<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">1d20h: ISDN Se3/0:15: Outgoing call id = 0x85F4, dsl 0<br />1d20h: ISDN Se3/0:15: process_pri_call(): call id 0x85F4, number 35293315, speed 0, call type VOICE, redialed? f, csm call? f, pdata? t<br />1d20h: callED type/plan overridden by call_decode<br />ld20h: did’t copy oct3a reason: not CALLER_NUMBER_IE<br />ld20h: building outgoing channel id for call nfas_int is 0 len is 0<br />ld20h: ISDN se3/0:15: TX -> INFOc sapi = 0 tei =0 ns = 19 nr = 19 i =<br />0x080200890504038090A31803A983811E0281837009803335323933333135<br />ld20h: SETUP pd = 8 callref = 0x0089<br />ld20h: Bearer capability i = 0x8090A3<br />ld20h: Channel id i = 0XA98381<br />ld20h: Progress Ind i =0x8183 – Origination address is non-ISDN<br />ld20h: called Party Number i = 0x80, ‘35293315’, Plan:unknown, Type:unknown<br />ld20h: ISDN Se3/0:15: RX <- RRr = 0 tei = 0 nr = 20<br />Id20h: ISDN se3/0:15: RX <- INFOc sapi = 0 tei =0 ns = 19 nr = 20 i = 0x080280895A08028286<br />ld20h: RELEASE_COMP pd = 8 callref = 0x8089<br />ld20h: Cause i = 0x8286 – Channel unacceptable<br />ld20h: ISDN Se3/0:15: TX -> RRr sapi = 0 tei =0 nr = 20<br />ld20h: ISDN se3/0:15: CCPRI_Releasecall(): bchan 1, call id 0x85F4, call type VOICE<br />ld20h: ccPRI_ReleaseChan released b_dsl 0 B_chan 1<br />ld20h: ISDN Se3/0:15: LIF_EVENT: ces/callid l/0x85F4 CALI__REJECTION<br />ld20h: ISDN Se3/0:15: LIF_EVENT: ces/callid 1/0x85F4 CALL_CLEARED<br />ld20h: ISDN Se3/0:15: received CALL_CLEARED call_id 0x85F4</td></tr>
</tbody></table>
<div style="margin-bottom: 10px; padding: 0px;">
A. Increase the ISDN T302 timer to allow more time for call setup.<br />B. Add the command isdn negotiate-bchan to the serial interface.<br />C. Add the command isdn contiguous-bchan to the serial interface.<br />D. Change the channel selection order from descending to ascending.</div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>B</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 3</div>
<div style="margin-bottom: 10px; padding: 0px;">
Refer to the exhibit. The Carmichael caller dials the site access code for Merrimack (6) followed by the four digit extension number of the destination phone (0124). If the call is going to go across the IP WAN, which action will have to be taken?</div>
<div style="margin-bottom: 10px; padding: 0px; text-align: center;">
<img alt="IPWAN_Routing.jpg" border="0" height="240" src="http://voicetut.com/images/CVoice/CallRouting/IPWAN_Routing.jpg" style="border: 0px; max-width: 800px;" width="600" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
A. Translate 60124 to 5125550124.<br />B. Strip the site access code and send four digits.<br />C. Strip the site access code and prepend 1512555.<br />D. Do nothing because the site access code matches the last five digits of the target number.<br />E. Strip the site access code, send four digits, then prepend the access code when it reaches the Merrimack gateway.</div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> B</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
The site access code (6) is just used to inform the originating gateway which gateway it needs to send traffic to. Therefore, after learning the traffic should be sent to Merrimack gateway, it trips off the site access code. Notice that the receiving gateway will receive “0124”, which is enough information to ring the phone plugged into it.</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 4</div>
<div style="margin-bottom: 10px; padding: 0px;">
Which path selection mechanism lets you choose either the even or odd channels first?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. hunt groups<br />B. trunk groups<br />C. tailend hopoff<br />D. Call Admission Control</div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> B</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
By using trunk groups, we can choose to use either the even or odd channels first with the command:</div>
<div style="margin-bottom: 10px; padding: 0px;">
<strong>hunt-scheme …. [even | odd ….] </strong>(notice: the full command is very long so I shorten it to the simplest form)</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 5</div>
<div style="margin-bottom: 10px; padding: 0px;">
When using CUBE, which two statements describe how media flow-through differs from media flow-around? (Choose two.)</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. Media flow-around provides address hiding by terminating both signaling and RTP streams.<br />B. Media flow-through terminates the signaling channel and the RTP streams flow directly between endpoints.<br />C. Media flow-around and media flow-through function in a similar manner, but media flow-around supports NAT traversal.<br />D. Media flow-through terminates the RTP streams but allows signaling to flow directly between endpoints.<br />E. Media flow-around terminates the signaling stream and allows RTP streams to flow directly between endpoints.<br />F. Media flow-through provides address hiding by terminating both signaling and RTP streams.</div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> E F</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
<strong>Media flow through</strong> and <strong>media flow around</strong> mode is supported on the Cisco Unified Border Element (CUBE). The CUBE is always involved in the call setup (signaling) portion of the call, but the media (RTP bearer stream) may flow through the CUBE or be routed around the platform.</div>
<div style="margin-bottom: 10px; padding: 0px;">
<strong>Media flow around</strong> allows the CUBE greater scalability in the number of calls that can be processed by one CUBE router.</div>
<div style="margin-bottom: 10px; padding: 0px;">
For <strong>Media flow through</strong> option, the media packets are passed through the CUBE, they will get terminated and re-originates with CUBE’s IP address and port number, so here we cannot find the original gateway’s ip address. This is one of the security feature in the CUBE. The default option is “media flow-through”.</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 6</div>
<div style="margin-bottom: 10px; padding: 0px;">
Refer to the IOS configuration in the exhibit. How will the next incoming call be routed?</div>
<table align="center" border="0" style="background-attachment: initial; background-clip: initial; background-image: initial; background-origin: initial; background-position: initial; background-repeat: initial; background-size: initial; border-collapse: collapse; border: 2px solid rgb(204, 204, 204); margin: 5px 0px 10px;"><tbody>
<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;"><strong>dial-peer voice 1 pots<br />translation-profile incoming in1</strong><div style="margin-bottom: 10px; padding: 0px;">
</div>
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<br class="spacer_" /></div>
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<br class="spacer_" /></div>
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<strong>trunk group 101</strong></div>
<div style="margin: 0px; padding: 0px;">
<strong>carrier-id 1642<br />hunt-scheme sequential even up<br />translation-profile incoming in1</strong></div>
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<strong><br /></strong></div>
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<strong>controller T1 1/0</strong></div>
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<strong>ds0-group 1 timeslots 1-24 type e&m-fgd<br />cas-custom 1<br />trunk-group 101</strong></div>
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<strong><br /></strong></div>
<div style="margin-bottom: 10px; padding: 0px;">
<strong>voice-port 1/0</strong></div>
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<strong>translation-profile incoming in1<br />trunk-group 101 1</strong></div>
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<strong><br /></strong></div>
<div style="margin-bottom: 10px; padding: 0px;">
<strong>voice service pots</strong></div>
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<strong>translation-profile incoming controller T1 1/0 in1</strong></div>
</td></tr>
</tbody></table>
<div style="margin-bottom: 10px; padding: 0px;">
A. The call will be routed to the longest idle channel.<br />B. The call will be routed to the least used channel.<br />C. The call will be routed to a random available channel.<br />D. The call will be routed to the next available channel, starting from channel 1, hunting up toward channel<br />E. The call will be routed to the next available channel, starting from channel 24, hunting down toward channel 1.</div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>E</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
In the configuration, we learn that the hunt-scheme sequential is used. It specifies the sequential search method for finding an available channel in a trunk group for outgoing calls. The syntax of this command is shown below:</div>
<div style="margin-bottom: 10px; padding: 0px;">
<strong>hunt-scheme sequential [both | even | odd [up | down] ]</strong></div>
<div style="margin-bottom: 10px; padding: 0px;">
<strong>Description:</strong></div>
<div style="margin-bottom: 10px; padding: 0px;">
+ <strong>both</strong>: Searches both even and odd numbered channels.<br />+ <strong>even</strong>: Searches for an idle even numbered channel. If no idle even numbered channel is available, an odd-numbered channel is sought.<br />+ <strong>odd</strong>: Searches for an idle odd numbered channel. If no idle odd numbered channel is available, an even-numbered channel is sought.</div>
<div style="margin-bottom: 10px; padding: 0px;">
+ <strong>up</strong>: Searches channels in ascending order based within a trunk group member.<br />+ <strong>down</strong>: Searches channels in descending order within a trunk group member.</div>
<div style="margin-bottom: 10px; padding: 0px;">
Notice that <strong>up </strong>& <strong>down </strong>parameters are used with <strong>both</strong>, <strong>even </strong>or <strong>odd</strong>.</div>
<div style="margin-bottom: 10px; padding: 0px;">
Therefore the command <strong>hunt-scheme sequential even up</strong> searches in ascending order for an even numbered idle channel starting with the trunk group member of highest precedence. I am not so sure but channel 24 will have highest precedence so the “hunt” begins from channel 24 down to channel 1. Therefore, E is the most suitable solution for this question.</div>
<div style="margin-bottom: 10px; padding: 0px;">
The <strong>cas-custom </strong>command is used to customize T1/CAS signaling parameters for a particular T1 channel group on a channelized T1 line.</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 7</div>
<div style="margin-bottom: 10px; padding: 0px;">
Site A uses three-digit internal numbers and remote Site B uses four-digit internal numbers. All calls to the PSTN are routed through Site B. What dial plan below best represents provision simplicity, assuming the NANP numbering plan?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. Translate all called numbers within Site A to four digits.<br />B. Translate all called numbers within Site B to three digits.<br />C. Translate all called numbers leaving Site A to ten digits.<br />D. Translate all called numbers at either site to ten digits.</div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>C</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
North American Numbering Plan (NANP) is designed around a 10-digit numbering plan:</div>
<div style="margin-bottom: 10px; padding: 0px; text-align: center;">
<img alt="NANP_tendigits.jpg" border="0" height="80" src="http://voicetut.com/images/CVoice/CallRouting/NANP_tendigits.jpg" style="border: 0px; max-width: 800px;" width="450" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
(Sometimes you will see it as NXX – NXXX – XXXX, which means that the first and fourth digits can’t be zero or one)</div>
<div style="margin-bottom: 10px; padding: 0px;">
It consists of 3-digit area codes and 7-digit telephone. For telephone numbers that are located within an area code, the PSTN uses a seven-digit dial plan numbers.</div>
<div style="margin-bottom: 10px; padding: 0px;">
Notice that “Site B uses four-digit internal numbers” means we need ten digits to access site B from an outside PSTN. Therefore, if people from Site A want to call people at site B and sometimes they just press 4 digits then the administrators should translate the called numbers to ten digits before leaving Site A.</div>
</div>
jcreatorhttp://www.blogger.com/profile/17926347588908332596noreply@blogger.com0tag:blogger.com,1999:blog-1786455060776668778.post-48217275861276385722015-07-16T17:12:00.002-07:002015-07-16T17:12:26.087-07:00Internet Telephony Service Provider<div class="content" style="background-color: white; color: #333333; font-family: Verdana, 'BitStream vera Sans', Tahoma, Helvetica, sans-serif; font-size: 12px; line-height: 17.3999996185303px; margin: 0px; overflow: hidden; padding: 5px 0px 0px 5px;">
<div style="margin-bottom: 10px; padding: 0px;">
Here you will find answers to CVoice – Internet Telephony Service Provider Questions</div>
<div style="margin-bottom: 10px; padding: 0px;">
</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 1</div>
<div style="margin-bottom: 10px; padding: 0px;">
You work as a network technician , study the exhibit carefully. The Acme Corp. uses H.323 to place calls to their supplier RR Industries. Acme also has a voice connection to an ITSP for long distance over a SIP network. Which configuration should Acme use to deploy the CUBE?</div>
<div style="margin-bottom: 10px; padding: 0px; text-align: center;">
<img alt="CUBE_Configuration.jpg" border="0" height="208" src="http://voicetut.com/images/CVoice/Internet_Telephony_Service_Provider/CUBE_Configuration.jpg" style="border: 0px; max-width: 800px;" width="505" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
A.</div>
<div style="margin-bottom: 10px; padding: 0px;">
service voice voip</div>
<div style="margin-bottom: 10px; padding: 0px;">
allow-connections h323 to h323</div>
<div style="margin-bottom: 10px; padding: 0px;">
allow-connections h323 to sip</div>
<div style="margin-bottom: 10px; padding: 0px;">
allow-connections sip to sip</div>
<div style="margin-bottom: 10px; padding: 0px;">
allow-connections sip to h323</div>
<div style="margin-bottom: 10px; padding: 0px;">
B.</div>
<div style="margin-bottom: 10px; padding: 0px;">
service voice voip</div>
<div style="margin-bottom: 10px; padding: 0px;">
allow-connections h323 to h323</div>
<div style="margin-bottom: 10px; padding: 0px;">
allow-connections h323 to sip</div>
<div style="margin-bottom: 10px; padding: 0px;">
C.</div>
<div style="margin-bottom: 10px; padding: 0px;">
voice service voip</div>
<div style="margin-bottom: 10px; padding: 0px;">
allow-connections h323 to h323</div>
<div style="margin-bottom: 10px; padding: 0px;">
allow-connections h323 to sip</div>
<div style="margin-bottom: 10px; padding: 0px;">
allow-connections sip to h323</div>
<div style="margin-bottom: 10px; padding: 0px;">
D.</div>
<div style="margin-bottom: 10px; padding: 0px;">
voice service voip</div>
<div style="margin-bottom: 10px; padding: 0px;">
allow-connections h323 to h323</div>
<div style="margin-bottom: 10px; padding: 0px;">
allow-connections h323 to sip</div>
<div style="margin-bottom: 10px; padding: 0px;">
<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>C</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
The Acme Corp connects to the ITSP via SIP Trunk and connects to RR industries via H.323. The Acme Corp itself uses H.323 so we have to enable protocol interworking with <strong>allow-connections</strong> commands:</div>
<div style="margin-bottom: 10px; padding: 0px;">
<strong>allow-connections h323 to h323</strong>: allow Acme Corp to communicate with RR industries (in both ways)</div>
<div style="margin-bottom: 10px; padding: 0px;">
<strong>allow-connections h323 to sip</strong>: allow Acme Corp to talk with ITSP (Acme Corp can talk and ITSP can hear but not vice versa)</div>
<div style="margin-bottom: 10px; padding: 0px;">
<strong>allow-connections sip to h323</strong>: allow ITSP to talk with Acme Corp (Acme Corp can hear and ITSP can talk but not vice versa)</div>
<div style="margin-bottom: 10px; padding: 0px;">
Notice that the configuration for H.323 and SIP interworking is unidirectional, thus if bidirectional interworking is required, you need to configure the mirror-matching statement as well.</div>
<div style="margin-bottom: 10px; padding: 0px;">
Acme Corp doesn’t use SIP so we don’t need to configure “allow-connections sip to sip”.</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 2</div>
<div style="margin-bottom: 10px; padding: 0px;">
H.323 is an umbrella Recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network. Which CUBE configuration will support H.323 protocol interworking and address hiding?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A.</div>
<div style="margin-bottom: 10px; padding: 0px;">
voice services voip</div>
<div style="margin-bottom: 10px; padding: 0px;">
h323 interworking</div>
<div style="margin-bottom: 10px; padding: 0px;">
media flow-around</div>
<div style="margin-bottom: 10px; padding: 0px;">
B.</div>
<div style="margin-bottom: 10px; padding: 0px;">
voice services h323 to h323</div>
<div style="margin-bottom: 10px; padding: 0px;">
h323 interworking</div>
<div style="margin-bottom: 10px; padding: 0px;">
media flow-through</div>
<div style="margin-bottom: 10px; padding: 0px;">
C.</div>
<div style="margin-bottom: 10px; padding: 0px;">
voice services voip</div>
<div style="margin-bottom: 10px; padding: 0px;">
allow-connections h323 to h323</div>
<div style="margin-bottom: 10px; padding: 0px;">
media flow-around</div>
<div style="margin-bottom: 10px; padding: 0px;">
D.</div>
<div style="margin-bottom: 10px; padding: 0px;">
voice service voip</div>
<div style="margin-bottom: 10px; padding: 0px;">
allow-connections h323 to h323</div>
<div style="margin-bottom: 10px; padding: 0px;">
<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>D</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
Address hiding is a security feature of the CUBE which will hide the IP address of the originating gateway. This feature is turn on by default so we don’t need to set it.</div>
<div style="margin-bottom: 10px; padding: 0px;">
A and B are not correct because the command “h323 interworking” doesn’t exist (moreover A uses “media flow-around” feature which will turn off the address hiding feature).</div>
<div style="margin-bottom: 10px; padding: 0px;">
C is not correct because it uses “media flow-around” feature too.</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 3</div>
<div style="margin-bottom: 10px; padding: 0px;">
Refer to the exhibit. The Acme Corp. is deploying a CUBE. As a component of protocol interworking between RR Industries and the ITSP, they need to configure at least two dial peers. When the IP WAN is functional, Acme Corp. wants to use 5-digit dialing to RR Industries. Which three dial peers will complete the configuration for Acme Corp.? (Choose three)</div>
<div style="margin-bottom: 10px; padding: 0px;">
<img alt="Dial_peer_Configuration.jpg" border="0" src="http://voicetut.com/images/CVoice/Internet_Telephony_Service_Provider/Dial_peer_Configuration.jpg" style="border: 0px; max-width: 800px;" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
A. dial-peer voice 50 voip<br />destination-pattern 50…<br />session protocol sipv2<br />session-target ipv4:192.168.100.100</div>
<div style="margin-bottom: 10px; padding: 0px;">
B. dial-peer voice 1000 voip<br />destination-pattern 51…<br />session-target ipv4:192.168.100.100</div>
<div style="margin-bottom: 10px; padding: 0px;">
C. dial-peer voice 91 voip<br />session protocol sipv2<br />destination-pattern 91T<br />session-target ipv4:10.1.100.1<br />dtmf-relay rtp-nte digit-drop h245-alphanumeric</div>
<div style="margin-bottom: 10px; padding: 0px;">
D. dial-peer voice 91 voip<br />destination-pattern 91T<br />session-target ipv4:10.1.100.1<br />dtmf-relay rtp-nte digit-drop h245-alphanumeric</div>
<div style="margin-bottom: 10px; padding: 0px;">
E. dial-peer voice 1000 voip<br />destination-pattern 51…<br />session-target ipv4:10.1.100.1</div>
<div style="margin-bottom: 10px; padding: 0px;">
F. dial-peer voice 50 voip<br />destination-pattern 50…<br />session-target ipv4:172.16.14.6</div>
<div style="margin-bottom: 10px; padding: 0px;">
<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>B C F</div>
</div>
jcreatorhttp://www.blogger.com/profile/17926347588908332596noreply@blogger.com0tag:blogger.com,1999:blog-1786455060776668778.post-80906379289849884692015-07-16T17:11:00.004-07:002015-07-16T17:11:45.154-07:00Call Signaling<div class="content" style="background-color: white; color: #333333; font-family: Verdana, 'BitStream vera Sans', Tahoma, Helvetica, sans-serif; font-size: 12px; line-height: 17.3999996185303px; margin: 0px; overflow: hidden; padding: 5px 0px 0px 5px;">
<div style="margin-bottom: 10px; padding: 0px;">
Here you will find answers to CVoice – Call Signaling Questions</div>
<div style="margin-bottom: 10px; padding: 0px;">
</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 1</div>
<div style="margin-bottom: 10px; padding: 0px;">
Which option is true concerning the MGCP call agent?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. acts only as a recorder of call details</div>
<div style="margin-bottom: 10px; padding: 0px;">
B. provides only call signaling and call setup</div>
<div style="margin-bottom: 10px; padding: 0px;">
C. manages all aspects of the call and voice stream</div>
<div style="margin-bottom: 10px; padding: 0px;">
D. monitors the quality of each call after setup</div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> B</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
MGCP Call Agent is a central control component to remotely control various devices. When the MGCP call agent exists in the network, calls are routed via route patterns on the Call Agent (Cisco Unified Communications Manager), not by dial peers on the gateway.</div>
<div style="margin-bottom: 10px; padding: 0px; text-align: center;">
<img alt="MGCP_CallAgent.jpg" border="0" height="300" src="http://voicetut.com/images/CVoice/CallSignaling/MGCP_CallAgent.jpg" style="border: 0px; max-width: 800px;" width="550" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
The messages sent between the voice gateway and the MGCP Call Agent are just used for call signaling and call setup only. In summary, the Call Agent will instruct the gateways what to do in each stage: receive dialed digits, find the destination gateway, send connection request… Finally, the Call Agent will allow gateways to establish RTP Streams with each other. Notice that the voice streams only flow between the two voice gateways, not to the Call Agent.</div>
<div style="margin-bottom: 10px; padding: 0px;">
At the conversation finishs (one of the endpoints goes on-hook), that gateway notifies the Call Agent and the Call Agent sends Delete Connection (DLCX) Requests for both gateways.</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 2</div>
<div style="margin-bottom: 10px; padding: 0px;">
At what point does the MGCP call agent release the setup of the call path to the residential gateways?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. after the call agent has been notified that an event occurred at the source residential gateway</div>
<div style="margin-bottom: 10px; padding: 0px;">
B. after the call agent has been notified of an event and has instructed the source residential gateway to create a connection</div>
<div style="margin-bottom: 10px; padding: 0px;">
C. does not release call path setup</div>
<div style="margin-bottom: 10px; padding: 0px;">
D. after the call agent has sent a connection request to both the source and destination and has relayed a modify-connection request to the source so that the source and destination can set up the call path</div>
<div style="margin-bottom: 10px; padding: 0px;">
E. after the call agent has forwarded session description protocol information to the destination from the source and has sent a modify connection to the destination and a create-connection request to the source</div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> D</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
Below is the call flow between two voice gateway through a MGCP Call Agent</div>
<div style="margin-bottom: 10px; padding: 0px; text-align: center;">
<img alt="MGCP_Call_Flow.jpg" border="0" height="600" src="http://voicetut.com/images/CVoice/CallSignaling/MGCP_Call_Flow.jpg" style="border: 0px; max-width: 800px;" width="620" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
The MGCP call agent releases the setup of the call path to the residential gateways when the conversation begins. After sending the Modify Connection (MDCX), the two gateways have enough information to start the conversation so the duty of the Call Agent finishs.</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 3</div>
<div style="margin-bottom: 10px; padding: 0px;">
Which three services are supported by CUBE when supporting H323-to-SIP calls? (Choose three)</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. SIP cause codes</div>
<div style="margin-bottom: 10px; padding: 0px;">
B. media flow-around</div>
<div style="margin-bottom: 10px; padding: 0px;">
C. media flow-through</div>
<div style="margin-bottom: 10px; padding: 0px;">
D. codec transparent support</div>
<div style="margin-bottom: 10px; padding: 0px;">
E. Transport Layer Security</div>
<div style="margin-bottom: 10px; padding: 0px;">
F. H.261, H.263, and H.264 video codecs</div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>C D E</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
<strong>Media flow through</strong> and <strong>media flow around</strong> mode is supported on the Cisco Unified Border Element (CUBE). The CUBE is always involved in the call setup (signaling) portion of the call, but the media (RTP bearer stream) may flow through the CUBE or be routed around the platform. <strong>Media flow through</strong> must be used to support many of the features available like IP address translation and IP address hiding. <strong>Media flow around</strong> allows the CUBE greater scalability in the number of calls that can be processed by one CUBE router.</div>
<div style="margin-bottom: 10px; padding: 0px;">
For “Media flow through” option, the media packets are passed through the CUBE, they will get terminated and re-originates with CUBE’s IP address and port number, so here we cannot find the original gateway’s ip address. This is one of the security feature in the CUBE. The default option is “media flow-through”.</div>
<div style="margin-bottom: 10px; padding: 0px;">
Use the “codec transparent” command to configure codec pass-through. Use this command to enable endpoint-to-endpoint codec negotiation</div>
<div style="margin-bottom: 10px; padding: 0px;">
without a Cisco UBE router -> D is correct.</div>
<div style="margin-bottom: 10px; padding: 0px;">
Transport Layer Security (TLS) is a security protocol that enables encrypted network communications. TLS on CUBE can be configured on a per-leg basis in order to allow a TLS to non-TLS SIP call. For example, CUBE uses IPSec in order to secure signaling and support calls from H.323 to SIP with the H.323 leg, while the SIP leg uses TLS -> E is correct</div>
<div style="margin-bottom: 10px; padding: 0px;">
The questions only let us choose 3 answers but I think<strong> B – media flow-around</strong> can be used for H323-to-SIP calls. We just can’t use media flow-around for SIP-to-SIP calls.</div>
<div style="margin-bottom: 10px; padding: 0px;">
For your information, H.323-to-SIP interworking is configured using the <strong>allow-connection h323 to SIP</strong> command. Then issue the <strong>allow-connections sip to h323</strong> command to enable SIP to H.323 calls.</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 4</div>
<div style="margin-bottom: 10px; padding: 0px;">
Which two are attributes of SCCP? (Choose two)</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. It is Cisco proprietary.</div>
<div style="margin-bottom: 10px; padding: 0px;">
B. It is a supervisory signaling protocol.</div>
<div style="margin-bottom: 10px; padding: 0px;">
C. It is classified as client/server architecture.</div>
<div style="margin-bottom: 10px; padding: 0px;">
D. SCCP devices are considered intelligent endpoints.</div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> A C</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
SCCP is the only Cisco-proprietary VoIP protocol currently in use. The purpose of SCCP protocol is to provide a signaling protocol between the Cisco Unified Communications Manager and Cisco IP phones. Similar to MGCP, SCCP is a client/server protocol -> A & C are correct.</div>
<div style="margin-bottom: 10px; padding: 0px;">
Supervisory signals involves the detection of changes to the status of a circuit (on-hook, off-hook, ringing). Any event causes a message to be sent to a Cisco UCM -> We can say SCCP is more than a supervisory signaling protocol because it tells the phone exactly what to do. From the on-hook, off-hook, buttons pressed, lamp on/off, through the prompt, key settings, and even the dialtone -> B is not correct.</div>
<div style="margin-bottom: 10px; padding: 0px;">
The beauty of SCCP is that it makes the endpoints very cheap comparing to the H.323 devices. The end stations (telephones) that use SCCP are</div>
<div style="margin-bottom: 10px; padding: 0px;">
called Skinny clients, which consume less processing overhead and they do not contain call control intelligence -> D is not correct.</div>
<div style="margin-bottom: 10px; padding: 0px; text-align: center;">
<img alt="SCCP_Topology.jpg" border="0" height="300" src="http://voicetut.com/images/CVoice/CallSignaling/SCCP_Topology.jpg" style="border: 0px; max-width: 800px;" width="600" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
SCCP devices (in this case the Cisco IP Phones) become “dump” devices when using this protocol because they have to ask the Unified Communications Manager for every action they need to do -> D is not correct.</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 5</div>
<div style="margin-bottom: 10px; padding: 0px;">
Refer to the exhibit. All IP phones are SCCP phones. Phone D makes an internal call to phone G. Which call setup signaling statement is true?</div>
<div style="margin-bottom: 10px; padding: 0px; text-align: center;">
<img alt="SCCP_Phones.jpg" border="0" height="420" src="http://voicetut.com/images/CVoice/CallSignaling/SCCP_Phones.jpg" style="border: 0px; max-width: 800px;" width="600" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
A. Phone D signals phone G directly. Call setup is handled by the phones.</div>
<div style="margin-bottom: 10px; padding: 0px;">
B. Phone D signals gateway A, which processes the call and signals phone G.</div>
<div style="margin-bottom: 10px; padding: 0px;">
C. Phone D signals gateway B, which processes the call and signals phone G.</div>
<div style="margin-bottom: 10px; padding: 0px;">
D. Phone D signals gatekeeper. The gatekeeper processes the call and signals phone G.</div>
<div style="margin-bottom: 10px; padding: 0px;">
E. Phone D signals the call agent. The call agent processes the call and signals phone G.</div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> E</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
This is a …weird and wrong question. Maybe the phone they want to ask here is Phone A, B or C because only these phones can use SCCP to communicate with the Call Agent. Phones D and E can’t use SCCP to talk with a H.323 Gateway.</div>
<div style="margin-bottom: 10px; padding: 0px;">
Phone A, B or C are SCCP Phones so they hand over the call control intelligence to the Call Agent and the Call Agent need to process the call and signals phone G before these phones can talk with each other.</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 6</div>
<div style="margin-bottom: 10px; padding: 0px;">
Which statement is true about MGCP?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. Call completion is always shared, with some intelligence on the endpoint, some on the call agent.</div>
<div style="margin-bottom: 10px; padding: 0px;">
B. Endpoints always take all actions to complete calls.</div>
<div style="margin-bottom: 10px; padding: 0px;">
C. Endpoints may act alone or cooperate with call agent to complete calls.</div>
<div style="margin-bottom: 10px; padding: 0px;">
D. Call agents order and direct each step of call completion for the endpoints.</div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>D</div>
</div>
jcreatorhttp://www.blogger.com/profile/17926347588908332596noreply@blogger.com0tag:blogger.com,1999:blog-1786455060776668778.post-41230182413077385012015-07-16T17:11:00.001-07:002015-07-16T17:11:12.221-07:00Analog Voice Ports<div class="content" style="background-color: white; color: #333333; font-family: Verdana, 'BitStream vera Sans', Tahoma, Helvetica, sans-serif; font-size: 12px; line-height: 17.3999996185303px; margin: 0px; overflow: hidden; padding: 5px 0px 0px 5px;">
<div style="margin-bottom: 10px; padding: 0px;">
Here you will find answers to CVoice – Analog Voice Port Questions</div>
<div style="margin-bottom: 10px; padding: 0px;">
</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 1</div>
<div style="margin-bottom: 10px; padding: 0px;">
Refer to the exhibit for IP addresses and telephone numbers. You are working with a customer opening a small sales office in Atlanta. You want the user in Atlanta to be able to dial into the PBX in New York over the IP WAN. The New York PBX uses ground start, a two-wire operation, and DTMF dialing. Choose the correct FXO port configuration commands for New York.</div>
<div style="margin-bottom: 10px; padding: 0px; text-align: center;">
<img alt="PBX_and_Phone.jpg" border="0" height="200" src="http://voicetut.com/images/CVoice/AnalogVoicePorts/PBX_and_Phone.jpg" style="border: 0px; max-width: 800px;" width="550" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
A.<br />voice-port 1/0/0<br />signal ground-start<br />operation 2-wire<br />dial-type dtmf</div>
<div style="margin-bottom: 10px; padding: 0px;">
B.<br />voice-port 1/1/1<br />destination 2015551212<br />signal ground-start<br />operation 2-wire<br />type 1<br />dial-type dtmf</div>
<div style="margin-bottom: 10px; padding: 0px;">
C.<br />voice port 1/0/0<br />session target ipv4:172.16.1.1<br />destination 2015551212<br />signal ground-start<br />operation 2-wire<br />dial-type dtmf</div>
<div style="margin-bottom: 10px; padding: 0px;">
D.<br />voice port 1/0/0<br />session target ipv4:172.16.1.1<br />source 2015551212<br />signal wink-start<br />operation 2-wire<br />dial-type dtmf</div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>A</div>
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Question 2</div>
<div style="margin-bottom: 10px; padding: 0px;">
Refer to the exhibit. Which configuration option will allow communication between a voice-enabled router and a PBX?</div>
<table border="0" style="background-attachment: initial; background-clip: initial; background-image: initial; background-origin: initial; background-position: initial; background-repeat: initial; background-size: initial; border-collapse: collapse; border: 2px solid rgb(204, 204, 204); margin: 5px 0px 10px;"><tbody>
<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;"><strong>Customer PBX System Parameters</strong><div style="margin-bottom: 10px; padding: 0px;">
</div>
<div style="margin-bottom: 10px; padding: 0px;">
<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
The calling PBX seizes the line by activating its M-lead.<br />The calling PBX toggles its M-lead on and off for a specific time.<br />The calling PBX receives the on/off signal and sends DTMF digits in the voice path<br />This is an older PBX so the on/off signal needs to be configured for the correct period of time<br />The voice path is 4-wires and the signaling path uses 2-wires</div>
</td></tr>
</tbody></table>
<div style="margin-bottom: 10px; padding: 0px;">
A. voice port 1/0/0<br />signaling wink-start<br />operation 4-wire<br />auto-cut-through<br />type 1</div>
<div style="margin-bottom: 10px; padding: 0px;">
B. voice port 1/0/0<br />signaling immediate-start<br />operation 4-wire<br />type 5</div>
<div style="margin-bottom: 10px; padding: 0px;">
C. voice port 1/0/0<br />signaling delay-start<br />auto-cut-through<br />operation 4-wire<br />type 3</div>
<div style="margin-bottom: 10px; padding: 0px;">
D. voice port 1/0/0<br />signaling wink-start<br />operation 4-wire<br />type 4</div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>A</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 3</div>
<div style="margin-bottom: 10px; padding: 0px;">
Examine the following PBX system parameters:</div>
<ul style="margin: 0px; padding: 0px 0px 10px;">
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">The calling side seizes the line by going off-hook on its E-lead and sends information as DTMF digits.</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">The voice path is 4-wires, and the voice enabled router is in another building from the PBX.</li>
</ul>
<div style="margin-bottom: 10px; padding: 0px;">
Select the correct set of commands to allow communication between a voice enabled router and a PBX.</div>
<div style="margin-bottom: 10px; padding: 0px;">
A.<br />voice port 1/0/0<br />signal immediate-start<br />operation 4-wire<br />type 2</div>
<div style="margin-bottom: 10px; padding: 0px;">
B.<br />voice-port 1/0/0<br />signal delay-dial<br />operation 4-wire<br />type 1</div>
<div style="margin-bottom: 10px; padding: 0px;">
C.<br />voice port 1/0/0<br />signal wink-start<br />operation 4-wire<br />type 3</div>
<div style="margin-bottom: 10px; padding: 0px;">
D.<br />voice port 1/0/0<br />signal immediate-start<br />operation 4-wire<br />type 4</div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>A</div>
</div>
jcreatorhttp://www.blogger.com/profile/17926347588908332596noreply@blogger.com0tag:blogger.com,1999:blog-1786455060776668778.post-10773304758402136552015-07-16T17:10:00.002-07:002015-07-16T17:10:11.619-07:00Voice Fundamentals<div class="content" style="background-color: white; color: #333333; font-family: Verdana, 'BitStream vera Sans', Tahoma, Helvetica, sans-serif; font-size: 12px; line-height: 17.3999996185303px; margin: 0px; overflow: hidden; padding: 5px 0px 0px 5px;">
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Here you will find answers to CVoice – Voice Fundamental Questions</div>
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</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 1</div>
<div style="margin-bottom: 10px; padding: 0px;">
Which two codes together make up the number that follows the E.164 recommendation numbering scheme? (Choose two)</div>
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A. country code<br />B. subscriber code<br />C. national destination code<br />D. provider code</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>A B</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
E.164 is an international numbering plan created by the International Telecommunication Union (ITU). Each number in the E.164 numbering plan</div>
<div style="margin-bottom: 10px; padding: 0px;">
contains the following components:</div>
<ul style="margin: 0px; padding: 0px 0px 10px;">
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">Country code (CC)</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">National destination code (NDC – optional)</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">Subscriber number (SN)</li>
</ul>
<div style="margin-bottom: 10px; padding: 0px;">
The CC consists of one, two or three digits. It is what we add in order to access different countries and often prefixed with a +</div>
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The NDC is the code we often call the area code.</div>
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The SN is for telephone numbering. It is given by your phone operator.</div>
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E.164 numbers are limited to a maximum length of 15 digits.</div>
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For example, the North American Numbering Plan E.164 is as follows:</div>
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<strong>1-602-555-1212</strong></div>
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+ 1: Country code</div>
<div style="margin-bottom: 10px; padding: 0px;">
+ 602555: National destination code (for North American Numbering Plan, 602 is called the area code while 555 is called Central Office Code)</div>
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+ 1212: Subscribe Number</div>
<div style="margin-bottom: 10px; padding: 0px;">
Answer C is also correct but just optional. E.164 Numbering Plan must have Country Code and Subscriber Code so A & B are the correct answers.</div>
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Question 2</div>
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Which statement is true about only out-of-band signaling?</div>
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A. A signaling bit is robbed from each frame.</div>
<div style="margin-bottom: 10px; padding: 0px;">
B. Signaling bits are sent in a special order in a dedicated signaling frame.</div>
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C. All signaling is directly associated with its corresponding voice frame.</div>
<div style="margin-bottom: 10px; padding: 0px;">
D. All voice packets carry their own signaling.</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> B</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
Out-of-Band signaling is telecommunication signaling exchange of information in order to control a telephone call. Out-of-Band signaling uses common channel signaling (CCS), that means signaling information is transmitted using a separate, dedicated signaling channel.</div>
<div style="margin-bottom: 10px; padding: 0px;">
Answers A C D are characteristics of Channel associated signaling (CAS) so they are not correct.</div>
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Question 3</div>
<div style="margin-bottom: 10px; padding: 0px;">
The D channel in ISDN is an example of which two signaling methods? (Choose two.)</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. CCS signaling</div>
<div style="margin-bottom: 10px; padding: 0px;">
B. out-of-band signaling</div>
<div style="margin-bottom: 10px; padding: 0px;">
C. in-band signaling</div>
<div style="margin-bottom: 10px; padding: 0px;">
D. CAS signaling</div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> A B</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
There are two types of ISDN lines: Basic Rate ISDN (BRI) and Primary Rate ISDN (PRI). Both BRI and PRI types have the same 64kbps D channel that is used for call supervision. This D channel is dedicated for signaling only and contains all the necessary signaling for establishing call between two end-points so it is a kind of CCS signaling and out-of-band signaling.</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 4</div>
<div style="margin-bottom: 10px; padding: 0px;">
In North America, which E&M signaling type is used most often for geographically separated equipment?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. Type I</div>
<div style="margin-bottom: 10px; padding: 0px;">
B. Type II</div>
<div style="margin-bottom: 10px; padding: 0px;">
C. Type III</div>
<div style="margin-bottom: 10px; padding: 0px;">
D. Type IV</div>
<div style="margin-bottom: 10px; padding: 0px;">
E. Type V</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>B</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
This information is quoted from <a href="http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a0080093f60.shtml" style="color: #2970a6; text-decoration: none;" target="_blank">http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a0080093f60.shtml</a></div>
<div style="margin-bottom: 10px; padding: 0px;">
<strong>E&M Type I</strong> – This is the most common interface in North America.</div>
<ul style="margin: 0px; padding: 0px 0px 10px;">
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">Type I uses two leads for supervisor signaling: E, and M.</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">During inactivity, the E-lead is open and the M-lead is connected to the ground.</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">The PBX (that acts as trunk circuit side) connects the M-lead to the battery in order to indicate the off-hook condition.</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">The Cisco router/gateway (signaling unit) connects the E-lead to the ground in order to indicate the off-hook condition.</li>
</ul>
<div style="margin-bottom: 10px; padding: 0px;">
<strong>E&M Type II</strong> – Two signaling nodes can be connected back-to-back.</div>
<ul style="margin: 0px; padding: 0px 0px 10px;">
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">Type II uses four leads for supervision signaling: E, M, SB, and SG.</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">During inactivity both the E-lead and M-lead are open.</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">The PBX (that acts as trunk circuit side) connects the M-lead to the signal battery (SB) lead connected to the battery of the signaling side in order to indicate the off-hook condition.</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">The Cisco router / gateway (signaling unit) connects the E-lead to the signal ground (SG) lead connected to the ground of the trunk circuit side in order to indicate the off-hook condition.</li>
</ul>
<div style="margin-bottom: 10px; padding: 0px;">
<strong>E&M Type III</strong> – This is not commonly used in modern systems.</div>
<ul style="margin: 0px; padding: 0px 0px 10px;">
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">Type III uses four leads for supervision signaling: E, M, SB, and SG.</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">During inactivity, the E-lead is open and the M-lead is set to the ground connected to the SG lead of the signaling side.</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">The PBX (that acts as trunk circuit side) disconnects the M-lead from the SG lead and connects it to the SB lead of the signaling side in order to indicate the off-hook condition.</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">The Cisco router / gateway (signaling unit) connects the E-lead to the ground in order to indicate the off-hook condition.</li>
</ul>
<div style="margin-bottom: 10px; padding: 0px;">
<strong>E&M Type IV</strong> – This is not supported by Cisco routers / gateways.</div>
<div style="margin-bottom: 10px; padding: 0px;">
<strong>E&M Type V</strong> – Type V is symmetrical and allows two signaling nodes to be connected back-to-back. This is the most common interface type used outside of North America.</div>
<ul style="margin: 0px; padding: 0px 0px 10px;">
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">Type V uses two leads for supervisor signaling: E, and M.</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">During inactivity the E-lead and M-lead are open.</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">The PBX ( that acts as trunk circuit side) connects the M-lead to the ground in order to indicate the off-hook condition.</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">The Cisco router / gateway (signaling unit) connects the E-lead to the ground in order to indicate off-hook condition.</li>
</ul>
<div style="margin-bottom: 10px; padding: 0px;">
Although above information specifies E&M Type 1 is the most commonly used interface in North America but this type generates significant delay in the signaling operation when transmitting between geographically separated equipment and affects voice signal quality (because of significant inductance and capacitance of the long wires) so Type 2 is often used instead.</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 5</div>
<div style="margin-bottom: 10px; padding: 0px;">
Which three are supervisory signals? (Choose three)</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. busy</div>
<div style="margin-bottom: 10px; padding: 0px;">
B. on hook</div>
<div style="margin-bottom: 10px; padding: 0px;">
C. off hook</div>
<div style="margin-bottom: 10px; padding: 0px;">
D. call waiting</div>
<div style="margin-bottom: 10px; padding: 0px;">
E. ring</div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>B C E</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
Supervisory signals involves the detection of changes to the status of a circuit. In other words, supervisory signaling is used to indicate the state of a circuit. Once these changes are detected, the supervisory circuit generates a predetermined response. There are three different types of supervisory signals, which are:</div>
<div style="margin-bottom: 10px; padding: 0px;">
1. On-hook</div>
<div style="margin-bottom: 10px; padding: 0px;">
2. Off-hook</div>
<div style="margin-bottom: 10px; padding: 0px;">
3. Ring</div>
<div style="margin-bottom: 10px; padding: 0px;">
When a telephone handset is in the cradle, the circuit is said to be on-hook. In on-hook state, the circuit is said to be open, thus preventing the current from flowing through the telephone.</div>
<div style="margin-bottom: 10px; padding: 0px;">
When the telephone handset is removed from the cradle, the circuit transitions to an off-hook state and there is a current flowing through the electrical loop. When the telephone network senses the off-hook state via the current flow, it provides a signal in the form of dial-tone that it is ready to accept the call. When making a call, the caller receives a ringback tone from the telephone switch, which alerts the caller that the telephone switch is sending ringing voltage to the called party. It is important to know that only the ringing that the recipient (the called party) hears is the supervisory signal; the ringback tone that the caller hears is simply a call-progress indicator and is not a supervisory signal.</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 6</div>
<div style="margin-bottom: 10px; padding: 0px;">
What is the approximate frequency range of human speech?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. 20 Hz to 20,000 Hz</div>
<div style="margin-bottom: 10px; padding: 0px;">
B. 40 Hz to 15,000 Hz</div>
<div style="margin-bottom: 10px; padding: 0px;">
C. 200 Hz to 9000 Hz</div>
<div style="margin-bottom: 10px; padding: 0px;">
D. 600 Hz to 5400 Hz</div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>C</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
Human speech ranges from 200 Hz to 9000 Hz but Nyquist cut the sampling frequency range to 4000 Hz to save bandwidth although this cut down the quality of voice too.</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 7</div>
<div style="margin-bottom: 10px; padding: 0px;">
What is the process of assigning audio amplitude to a unique digital code word?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. linear prediction</div>
<div style="margin-bottom: 10px; padding: 0px;">
B. encoding</div>
<div style="margin-bottom: 10px; padding: 0px;">
C. sampling</div>
<div style="margin-bottom: 10px; padding: 0px;">
D. quantization</div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> D</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
Quantization is the process of assigning a value from the voltage range based on the amplitude of each audio sample. Notice that the Voltage values are not evenly spaced. The spaces near the horizontal line are much closer than the ones at the two ends. This helps our ears distinguish common sounds more easily.</div>
<div style="margin-bottom: 10px; padding: 0px; text-align: center;">
<img alt="quantizing_voice.jpg" height="422" src="http://voicetut.com/images/CVoice/VoiceFundamentals/quantizing_voice.jpg" style="border: 0pt none; max-width: 800px;" width="460" /></div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 8</div>
<div style="margin-bottom: 10px; padding: 0px;">
What is the E.164 standard?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. private numbering plan</div>
<div style="margin-bottom: 10px; padding: 0px;">
B. national numbering plan</div>
<div style="margin-bottom: 10px; padding: 0px;">
C. dial plan</div>
<div style="margin-bottom: 10px; padding: 0px;">
D. international public telecommunications numbering plan</div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> D</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
E.164 is an ITU-T recommendation which defines the international public telecommunication numbering plan used in the PSTN and some other data networks.</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 9</div>
<div style="margin-bottom: 10px; padding: 0px;">
For the following items, which is the most common E&M type used outside North America?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. Type IV</div>
<div style="margin-bottom: 10px; padding: 0px;">
B. Type I</div>
<div style="margin-bottom: 10px; padding: 0px;">
C. Type II</div>
<div style="margin-bottom: 10px; padding: 0px;">
D. Type III</div>
<div style="margin-bottom: 10px; padding: 0px;">
E. Type V</div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> E</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
Read question 4</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 10</div>
<div style="margin-bottom: 10px; padding: 0px;">
A new business in Great Britain needs to have a PSTN connection that will handle a maximum of 30 inbound and outbound calls at any given time. The customer only has one slot available on the designated PSTN router. Which digital line type will you recommend?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. ISDN T1 PRI</div>
<div style="margin-bottom: 10px; padding: 0px;">
B. QSIG</div>
<div style="margin-bottom: 10px; padding: 0px;">
C. ISDN E1 PRI</div>
<div style="margin-bottom: 10px; padding: 0px;">
D. ISDN BRI</div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> C</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
The ISDN E1 PRI has 32 timeslots (channels). Each timeslot is 8 bits and has a data rate of 64,000 bits/second. Timeslot 0 is used for frame synchronization and alarms. Timeslot 16 is used for signaling so we can use 30 timeslots to carry calls.</div>
<div style="margin-bottom: 10px; padding: 0px;">
ISDN T1 PRI only has 24 timeslots and can not support 30 simultaneous calls.</div>
<div style="margin-bottom: 10px; padding: 0px;">
QSIG is just an ISDN based signaling protocol for signaling.</div>
</div>
jcreatorhttp://www.blogger.com/profile/17926347588908332596noreply@blogger.com0tag:blogger.com,1999:blog-1786455060776668778.post-9432673657939707462015-07-16T17:06:00.003-07:002015-07-16T17:06:33.040-07:00Share your CCNA Voice Experience<div class="content" style="background-color: white; color: #333333; font-family: Verdana, 'BitStream vera Sans', Tahoma, Helvetica, sans-serif; font-size: 12px; line-height: 17.3999996185303px; margin: 0px; overflow: hidden; padding: 5px 0px 0px 5px;">
<div class="pinkandbold" style="color: magenta; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Please share with us your experience after taking the CCNA Voice 640-460 exam, your materials, the way you learned, your recommendations…</div>
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Your posts are warmly welcome!</div>
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jcreatorhttp://www.blogger.com/profile/17926347588908332596noreply@blogger.com0tag:blogger.com,1999:blog-1786455060776668778.post-64248385637508446892015-07-16T17:06:00.000-07:002015-07-16T17:06:09.148-07:00Other Questions<div class="content" style="background-color: white; color: #333333; font-family: Verdana, 'BitStream vera Sans', Tahoma, Helvetica, sans-serif; font-size: 12px; line-height: 17.3999996185303px; margin: 0px; overflow: hidden; padding: 5px 0px 0px 5px;">
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Here you will find answer to other questions in the CCNA Voice 640-460 Exam</div>
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</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 1</div>
<div style="margin-bottom: 10px; padding: 0px;">
A SIP Trunk is a logical connection between an IP PBX and a Service Provider’s application servers that allows voice over IP traffic to be exchanged between the two. When a call is placed from an internal phone to an external number, the PBX sends the necessary information to the SIP Trunk provider who establishes the call to the dialed number and acts as an intermediary for the call. All signaling and voice traffic between the PBX and the provider is exchanged using SIP and RTP protocol packets over the IP network. Which two statements about SIP trunk are true?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. A SIP trunk configuration is mandatory for a UC500 device.<br />B. A SIP trunk is needed for internet access<br />C. A SIP trunk is needed only for voice if you are planning on using VoIP through a service provider.<br />D. A SIP trunk is not supported in a keyswitch configuration.</div>
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<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>C D</div>
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Question 2</div>
<div style="margin-bottom: 10px; padding: 0px;">
What is the default VTP mode on Cisco switches?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. Master<br />B. Client<br />C. Backup<br />D. Server</div>
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<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> D</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 3</div>
<div style="margin-bottom: 10px; padding: 0px;">
What are the three VTP modes on Cisco switches?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. Master<br />B. Client<br />C. Server<br />D. Transparent</div>
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<br class="spacer_" /></div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> B C D</div>
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Question 4</div>
<div style="margin-bottom: 10px; padding: 0px;">
What protocol needs to be enabled on an ATA if a fax machine is connected to the ATA?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. MGCP<br />B. SCCP<br />C. H323<br />D. SIP</div>
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<br class="spacer_" /></div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> C</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 5</div>
<div style="margin-bottom: 10px; padding: 0px;">
When would you require a voice gateway? (Choose two)</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. When you conned a branch location to IP WAN<br />B. When you connect a branch location using VoIP to the PSTN<br />C. When you connect a Cisco Unified Communications Manager to a LAN<br />D. When you connect a Cisco Unified Communications network to a PBX</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>B D</div>
</div>
jcreatorhttp://www.blogger.com/profile/17926347588908332596noreply@blogger.com0tag:blogger.com,1999:blog-1786455060776668778.post-81278666007976102902015-07-16T17:05:00.002-07:002015-07-16T17:05:38.507-07:00Voice Over IP<div class="content" style="background-color: white; color: #333333; font-family: Verdana, 'BitStream vera Sans', Tahoma, Helvetica, sans-serif; font-size: 12px; line-height: 17.3999996185303px; margin: 0px; overflow: hidden; padding: 5px 0px 0px 5px;">
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Here you will find answers to Voice Over IP Questions</div>
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</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 1</div>
<div style="margin-bottom: 10px; padding: 0px;">
Which type of delay can lead to jitter in a voice network?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. Propagation delay<br />B. Serialization delay<br />C. CODEC delay<br />D. Queuing delay</div>
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<br class="spacer_" /></div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> D</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
Jitter is the variation in the arrival of voice packets. For example, the first voice packet of a conversation might take 50 ms to reach a destination while the second voice packet might take 60 ms. There is 10 ms of delay variation (jitter) between these packets. The varying arrival time of the packets can cause gaps in the re-creation and playback of the voice signal. These gaps are undesirable and annoy the listener. For example, if the speaker says “Enjoy your life” then the listener will hear “Ennnnnnjoy yooooour liiiiiiife”.</div>
<div style="margin-bottom: 10px; padding: 0px;">
Queuing delay (how long a packet waits in a router’s interface queue) is variable because it depends on how many packets are currently in the queue. Therefore queuing delay is the main reason leading to jitter in a VoIP network.</div>
<div style="margin-bottom: 10px; padding: 0px;">
Other delays (propagation, serialization, CODEC) are fixed and predictable delays.</div>
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+ Propagation: The time it takes a packet to traverse a link.<br />+ Serialization: The insertion of bits onto a link.<br />+ CODEC: The time for translating the audio signal into a digital signal.</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 2</div>
<div style="margin-bottom: 10px; padding: 0px;">
IP phone A places a call to IP phone B. How many RTP streams are required for the call to be successfully completed?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. 1<br />B. 2<br />C. 4<br />D. 6</div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> B</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
Notice that RTP streams are one-way. If you are having a two-way conversation, the devices will establish dual RTP streams, one in each direction</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 3</div>
<div style="margin-bottom: 10px; padding: 0px;">
Refer to the exhibit.</div>
<div style="margin-bottom: 10px; padding: 0px; text-align: center;">
<img alt="dial_peers.jpg" border="0" height="222" src="http://voicetut.com/images/CCNA_Voice/VoiceOverIP/dial_peers.jpg" style="border: 0px; max-width: 800px;" width="600" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
The Acme Corporation needs assistance in configuring their PSTN voice gateway. Which two dial peers will correctly route calls to emergency services? (Choose two)</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. dial-peer voice 1 pots<br />destination-pattern 9911<br />port 1/0/0</div>
<div style="margin-bottom: 10px; padding: 0px;">
B. dial-peer voice 911 pots<br />destination-pattern 911<br />forward-digits 3<br />port 1/0/1</div>
<div style="margin-bottom: 10px; padding: 0px;">
C. dial-peer voice 9911 pots<br />destination-pattern 9911<br />forward-digits all<br />port 1/0/0</div>
<div style="margin-bottom: 10px; padding: 0px;">
D. dial-peer voice 2 pots<br />destination-pattern 911<br />forward-digits 3<br />port 1/0/1</div>
<div style="margin-bottom: 10px; padding: 0px;">
E. dial-peer voice 1 pots<br />destination-pattern 9911<br />prefix 911<br />port 1/0/0</div>
<div style="margin-bottom: 10px; padding: 0px;">
F. dial-peer voice 2 pots<br />destination-pattern 911<br />forward-digits all<br />port 1/0/0</div>
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<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> E F</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
The first time I read this question, I think the Local PSAP (Public Service Answering Point) can accept both 911 and 9911 but it is not true. The Local PSAP only accept 911 so the duty of the administrator is to configure the gateway in order to support both 911 and 9911 numbers. To do this, we need two dial-peers, one for 911 and another for 9911.</div>
<div style="margin-bottom: 10px; padding: 0px;">
But keep in mind that our outgoing dial-peer (port FXO 1/0/0) is a POTS dial-peer so the matched digits of this dial-peer will get stripped so we need to use the <strong>forward-digits all</strong> or <strong>forward-digits 3</strong> (for 911 pattern) or <strong>prefix 911</strong> (for 9911 pattern) to keep the called number. Therefore only E and F are correct.</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 4</div>
<div style="margin-bottom: 10px; padding: 0px;">
Approximately what percentage of voice packets can be dropped before voice quality becomes poor?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. 1 to 2%<br />B. 15%<br />C. 5 to 10%<br />D. Less than or equal to 1%</div>
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<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> D</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
Packet loss describes an error condition in which data packets appear to be transmitted correctly at one end of a connection, but never arrive at the other. This might be because:</div>
<ul style="margin: 0px; padding: 0px 0px 10px;">
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">network conditions are poor and the packet became damaged in transit</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">The packet was deliberately dropped at a router because of internet congestion.</li>
</ul>
<div style="margin-bottom: 10px; padding: 0px;">
The 1% threshold is just an estimate. Some documents say that even with 1% packet loss can “significantly degrade” a VOIP call using G.711 or G.729 codec. But “1% or less” is the best answer for this question.</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 5</div>
<div style="margin-bottom: 10px; padding: 0px;">
How does LLQ help ensure that voice quality is maintained in a converged network?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. LLQ allocates minimum bandwidth guaranteed to voice traffic.<br />B. LLQ allocates a priority queue to voice traffic at a guaranteed rate.<br />C. LLQ allocates a priority queue and a minimum guaranteed bandwidth queue for voice.<br />D. LLQ ensures that all traffic is treated fairly and hence voice traffic is not severely impacted.</div>
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<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>B</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
Low-latency queuing (LLQ) is used to give specific traffic classes higher priority when transmitting on the router’s WAN interface. Low Latency Queuing allows delay-sensitive data such as voice to be dequeued and sent first (before packets in other queues are dequeued), giving delay-sensitive data preferential treatment over other traffic.</div>
<div style="margin-bottom: 10px; padding: 0px;">
The bandwidth given to an LLQ priority queue (PQ) is both the guaranteed minimum and policed maximum. This helps prevent the queue starvation that occurs with PQ.</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 6</div>
<div style="margin-bottom: 10px; padding: 0px;">
In which two situations would a voice gateway be required? (Choose two)</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. To connect a corporate or branch location to an IP WAN<br />B. To connect a corporate or branch location using VoIP to the PSTN<br />C. To connect a Cisco Unified Communications Manager to a LAN<br />D. To connect a Cisco Unified Communications network to a PBX<br />E. To connect a corporate or branch location to a MAN</div>
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<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>B D</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 7</div>
<div style="margin-bottom: 10px; padding: 0px;">
What protocol is used to monitor and provide control information about the quality of an RTP session?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. UDP<br />B. RTP<br />C. NTP<br />D. RTCP</div>
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<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> D</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
RTCP is used to monitor and provide control information about the quality of RTP streams but notice that RTCP only provides feedback on the quality of the transmission link. It does not make any guarantees concerning quality of service.</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 8</div>
<div style="margin-bottom: 10px; padding: 0px;">
Which three are components of a dial plan? (Choose three)</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. Call legs<br />B. Endpoint addressing<br />C. centralized control<br />D. Call coverage<br />E. Digit manipulation<br />F. Decentralized control</div>
<div style="margin-bottom: 10px; padding: 0px;">
<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> B D E</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
We should understand what a dial plan is before talking about its components. In short, a dial plan is a collection of rules the call-processing agent uses to route calls. Below lists the components of a dial plan:</div>
<div style="margin-bottom: 10px; padding: 0px;">
<strong>Endpoint addressing</strong> is the addressing scheme that is used to reach voice endpoints. For example, a company numbering plan might use four-digit extensions at each location and a three-digit site code. To call a phone at your own location, you would dial the four-digit extension. To call a phone at a remote company location, you would dial the site code and the extension.</div>
<div style="margin-bottom: 10px; padding: 0px;">
<strong>Call coverage:</strong> Special groups of devices can be created to handle incoming calls for a certain service according to different rules, avoiding dropped calls. For example: top-down, circular hunt, longest idle, or broadcast groups are popular ones that you will see while learning CCNA Voice.</div>
<div style="margin-bottom: 10px; padding: 0px;">
<strong>Digit manipulation:</strong> Digits can be manipulated prior to or after a routing decision has been made. In some cases, it is necessary to manipulate the dialed string before routing the call, for example, when you are rerouting over the PSTN a call originally dialed using the on-net access code, or when you are expanding an abbreviated code (such as 0 for the operator) to an extension.</div>
<div style="margin-bottom: 10px; padding: 0px;">
Other two components of a dial peer are:</div>
<div style="margin-bottom: 10px; padding: 0px;">
<strong>Calling privileges </strong>(or COR – class of service): Different groups of devices can be assigned to different classes of service, by granting or denying access to certain destinations or resources. For example, Employee phones might be allowed to reach only internal and local PSTN destinations, while Executive phones could have unrestricted PSTN access. The calling privileges assigned to a device are typically called class of service. In a Cisco voice<br />gateway, class of service is implemented by assigning Class of Restrictions (COR) to dial peers.</div>
<div style="margin-bottom: 10px; padding: 0px;">
<strong>Path selection:</strong> Depending on the calling device, different paths can be selected to reach the same destination. Moreover, a secondary path can be used when the primary path is not available (for example, a call can be transparently rerouted over the PSTN during an IP WAN failure).</div>
<div style="margin-bottom: 10px; padding: 0px;">
(Reference: CCVP – Implementing Cisco Voice Gateways and Gatekeeper & CVoice v6.0 Module 4 Lesson 1)</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 9</div>
<div style="margin-bottom: 10px; padding: 0px;">
Refer to the exhibit.</div>
<table align="center" border="1" cellpadding="3" style="background-attachment: initial; background-clip: initial; background-image: initial; background-origin: initial; background-position: initial; background-repeat: initial; background-size: initial; border-collapse: collapse; border: 2px solid rgb(204, 204, 204); margin: 5px 0px 10px;"><tbody>
<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">A</td><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">dial-peer voice 6000 voip</td></tr>
<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">B</td><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">destination-pattern 19..</td></tr>
<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">C</td><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">session protocol sipv2</td></tr>
<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">D</td><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">session target ipv4:10.19.153.2</td></tr>
<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">E</td><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">dtmf-relay sip-notify</td></tr>
<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">F</td><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">codec g729ulaw</td></tr>
<tr><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">G</td><td style="border: 1px solid rgb(204, 204, 204); padding: 3px 10px; vertical-align: top;">no vad</td></tr>
</tbody></table>
<div style="margin-bottom: 10px; padding: 0px;">
The configuration shows a dial peer that points to Cisco Unity Express. Which line of configuration is incorrect?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. B<br />B. C<br />C. D<br />D. E<br />E. F<br />F. G</div>
<div style="margin-bottom: 10px; padding: 0px;">
<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>E</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
We don’t have <strong>G.729</strong>ulaw, just <strong>G.711</strong>ulaw. G729 has 3 annexes that are G.729a, G.729b and G.729ab.</div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 10</div>
<div style="margin-bottom: 10px; padding: 0px;">
Refer to the exhibit.</div>
<div style="margin-bottom: 10px; padding: 0px; text-align: center;">
<img alt="call_legs.jpg" border="0" height="193" src="http://voicetut.com/images/CCNA_Voice/VoiceOverIP/call_legs.jpg" style="border: 0px; max-width: 800px;" width="600" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
How many discrete call legs are needed to set up a call between the POTS phone attached to router 1 and the phone in the PSTN?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. 3<br />B. 4<br />C. 6<br />D. 7<br />E. 8<br />F. 10</div>
<div style="margin-bottom: 10px; padding: 0px;">
<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>B</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
We need four call legs as shown below</div>
<div style="margin-bottom: 10px; padding: 0px; text-align: center;">
<img alt="dial_peer_types.jpg" border="0" height="209" src="http://voicetut.com/images/CCNA_Voice/VoiceOverIP/dial_peer_types.jpg" style="border: 0px; max-width: 800px;" width="600" /></div>
<div class="ccnaquestionsnumber" style="color: #ff3300; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Question 11</div>
<div style="margin-bottom: 10px; padding: 0px;">
Refer to the exhibit.</div>
<div style="margin-bottom: 10px; padding: 0px; text-align: center;">
<img alt="match_dial_peer.jpg" border="0" height="186" src="http://voicetut.com/images/CCNA_Voice/VoiceOverIP/match_dial_peer.jpg" style="border: 0px; max-width: 800px;" width="600" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
Which inbound dial peer on CMERouter1 will be matched when phone extension 1234 places a call to 2010?</div>
<div style="margin-bottom: 10px; padding: 0px;">
A. Voip dial peer 30<br />B. Default dial peer 30<br />C. None, which will cause the call to drop<br />D. Default dial peer</div>
<div style="margin-bottom: 10px; padding: 0px;">
<br class="spacer_" /></div>
<div style="margin-bottom: 10px; padding: 0px;">
<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> D</div>
<div class="ccnaexplanation" style="color: #66ff00; font-weight: bold; margin-bottom: 10px; padding: 0px;">
Explanation</div>
<div style="margin-bottom: 10px; padding: 0px;">
For CMERouter1 the “dial-peer voice 30 voip” will be matched for the outbound dial peer, not inbound one. When there is no dial-peer matched, the router will use the default dial peer.</div>
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Question 12</div>
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Refer to the exhibit.</div>
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<img alt="dial_peer_types.jpg" border="0" height="209" src="http://voicetut.com/images/CCNA_Voice/VoiceOverIP/dial_peer_types.jpg" style="border: 0px; max-width: 800px;" width="600" /></div>
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Which two types of dial peers are needed to complete this call end-to-end? (Choose two)</div>
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A. Serial dial peer<br />B. PSTN dial peer<br />C. POTS dial peer<br />D. Network dial peer<br />E. VoIP dial peer</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>C E</div>
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Question 13</div>
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What is the relationship between a call leg and a dial peer?</div>
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A. A call leg is a virtual connection to set up a call whereas a dial peer is a physical connection to complete an end-to-end call.<br />B. The call leg and the dial peers are both logical connections used to complete an end-to-end call.<br />C. A call leg is a virtual connection that is set up and torn down before the dial peer is established.<br />D. The call leg and the dial peer are both physical connections used to complete an end-to-end call.</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>B</div>
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Question 14</div>
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Which type of voice port will be most cost effective to allow the gateway to terminate two circuits from the PSTN or a PBX?</div>
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A. FXO<br />B. FXS<br />C. PRI T1<br />D. E1<br />E. E&M<br />F. BRI<br />G. CAS T1</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> E</div>
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Explanation</div>
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For PSTN and PBX connection, we need to use an analog interface type. E&M signaling is designed to connect directly to a PBX system that also supports E&M interfaces. Many PBX brands have E&M analog trunk cards that can operate as either the trunk circuit side or the signaling unit side and Cisco gateway does support E&M interfaces.</div>
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Question 15</div>
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Which of the following is selected first for an incoming dial peer?</div>
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A. Answer-address<br />B. incoming called-number<br />C. destination-pattern<br />D. pots port</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> B</div>
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Explanation</div>
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First, the gateway attempts to match the called number with the<strong> incoming called-number</strong>. If no match is found, the router or gateway attempts to match the calling number of the call set-up request with the <strong>answer-address</strong> of each dial-peers. If no match is found, it attempts to match the calling number of the call set-up request to the <strong>destination-pattern</strong> of each dial-peer.</div>
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Notice that these steps are just applied for inbound dial peer.</div>
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Question 16</div>
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Which protocol provides VoIP packet sequence numbering?</div>
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A. IP<br />B. TCP<br />C. UDP<br />D. RTP</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>D</div>
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Explanation</div>
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The RTP protocol provides end-to-end network functions and delivery services for delay-sensitive, real-time data, such as voice and video. It runs on top of UDP and provides these services:</div>
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<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">Payload-type identification</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">Sequence numbering</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">Time stamping</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">Delivery monitoring</li>
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Question 17</div>
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Identify the VoIP network component that provides CAC, bandwidth control and management, and address translation.</div>
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A. Gateway<br />B. Gatekeeper<br />C. MCU<br />D. Call agent</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>B</div>
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Explanation</div>
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A gatekeeper can perform these tasks:</div>
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<strong>Address translation:</strong> The gatekeeper translates alias addresses (e.g., E.164 telephone numbers) to Transport Addresses, using a translation table that is updated using Registration messages and other means.</div>
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<strong>Bandwidth control:</strong> The gatekeeper controls how much bandwidth a terminal may use. The gatekeeper provides the above functions for terminals and gateways that have registered with it.</div>
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<strong>Bandwidth management</strong>: Limits the number of concurrent accesses to IP internetwork resources (gatekeeper-based CAC for bandwidth management) (CAC: Call Admission Control).</div>
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Question 18</div>
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Which three of the following are appropriate solutions to address latency issues in a VoIP network? (Choose 3)</div>
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A. Use dejitter buffers<br />B. Increase bandwidth<br />C. Fragment data packets<br />D. Prioritize voice packets</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>B C D</div>
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Explanation</div>
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Notice that buffers give smoother audio playout but they does increase latency in VoIP network.</div>
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Question 19</div>
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Which three headers are compressed by cRTP? (Choose 3)</div>
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A. Data link<br />B. IP<br />C. UDP<br />D. RTP</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer: </span>B C D</div>
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Explanation</div>
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Compressed Real-Time Transport Protocol (cRTP) compresses IP/UDP/RTP headers on low-speed serial links. We shouldn’t use cRTP on any high-speed interfaces as the price of CPU utilization is higher than the bandwidth savings.</div>
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Question 20</div>
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Which of the following best describes a function of RTCP?</div>
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A. RTCP provides encryption, message authentication and integrity, and anti-replay service for voice streams<br />B. RTCP uses even-numbered UDP ports in the range 16,384-32,767 to transport voice payloads<br />C. RTCP provides out-of-band control information for an RTP flow<br />D. RTCP caches an RTP packet’s Layer 3 and Layer 4 headers in the routers at each end of a link, resulting in lower bandwidth demand for subsequent RTP packets</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> C</div>
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Explanation</div>
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While the Real-time Transport Control Protocol (RTCP) sounds very inportant, its primary job is just statistics reporting, which includes</div>
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<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">Packet count</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">Packet Delay</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">Packet Loss</li>
<li style="list-style-position: inside; margin: 0px; padding: 0px 0px 0px 20px;">Jitter (delay variations)</li>
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These types of information are useful but not as important as the actual RTP audio streams. Keep this in mind to configure RTCP & RTP streams correctly in the future.</div>
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Question 21</div>
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Which two of the following VoIP gateway platforms are considered to be Integrated Services Routers (ISRs)? (Choose two)</div>
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A. Cisco 2600XM Series<br />B. Cisco 2800 Series<br />C. Cisco 3700 Series<br />D. Cisco 3800 Series</div>
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<span class="ccnacorrectanswers" style="color: #3333cc; font-weight: bold;">Answer:</span> B D</div>
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Explanation</div>
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We can hardly find a complete definition for the Integrated Services Routers but you can understand ISR as following:</div>
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“An ISR integrates other network features into the router other than just routing features. Used mostly in small offices on ADSL lines, they offer things like VPN, firewall, and encryption services.”</div>
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or another definition:</div>
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“First, the ISR routers are devices with a low-performance CPU when comparing them to the usual workstation/server processors from Intel or AMD.</div>
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Second, they are, as their name suggests it, “integrated services routers”, i.e. universal devices capable of performing many diverse networking functions, and that is true. However, even if a device can provide a particular service, it does not mean that it has unlimited power for providing it, and also if a device supports various features, it does not necessarily mean that you can have all of them turned on and expect that they all will perform well under a high load. The ISR routers are very flexible, however, they are still considered to be, at least from the throughput point of view, low-end routers. Their strength is the versatility, not the raw throughput.”</div>
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<strong>Some benefits of Integrated Services Routers:</strong></div>
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1. The ISRs are more cost effective than their legacy equivalents, particularly when the network requirements map to an existing bundle.<br />2. The ISRs are faster (up to five times) and can handle quite a bit more memory than the legacy platforms. The base configurations also have more memory.<br />3. The ISRs are designed with the ability to run multiple concurrent services (FW, NAT, IDS, QoS, etc.) at wire-speed.<br />4. All the ISRs have TWO built-in LAN connections – FE or GE.<br />5. All the ISRs have an embedded HW VPN accelerator – It is always included, it is just a matter of buying a VPN enabled image to turn it on. If that is not fast enough, a VPN AIM can be added to further enhance VPN performance.<br />6. The HWIC enabled slots provide an impressive 400Mbps of dedi-cated bandwidth (the old WICs provided up to 8Mbps). This is great news for LAN uplinks and Ethernet Switch HWICs. The NME slots offer up to 1.2Gbps per module (the standard NM was only 600Mbps).<br />7. The EVM slot offers high density digital/analog voice ports.<br />8. All the ISRs with voice support have on-board DSP slots. There is no need to use a NM slot for a network module with DSPs for voice applications anymore – the on-board DSP slots can provide enough DSP resources for most common requirements.<br />9. All the ISRs with voice support can provide voice mail functionality with CUE (AIM and/or NM). CUE was not supported on the 1700 family.<br />10. All the ISRs that support voice can provide in-line power to Ethernet switch ports via a HWIC-ESW-POE or a NM-ESW-PWR (optional AC-IP power supply is required for in-line power).<br />11. Most ISRs provide some option for power supply redundancy. The 2811, 2821, 2851 and 3825 have a RPS connector and the 3845 can take a built-in redundant power supply.<br />12. For investment protection, the ISRs support most of the existing WICs, VICs, VWICs and NM modules (check the datasheets for de-tails).<br />13. SDM (Security Device Manager), included on all ISRs.</div>
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