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300-070 Implementing Cisco IP Telephony and Video, portion 1 (CIPTV1)

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300-070 exam Dumps Source : Implementing Cisco IP Telephony and Video, portion 1 (CIPTV1)

Test Code : 300-070
Test appellation : Implementing Cisco IP Telephony and Video, portion 1 (CIPTV1)
Vendor appellation : Cisco
free pdf : 211 actual Questions

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Cisco Implementing Cisco IP Telephony

Cisco IP Telephony (working towards) | killexams.com actual Questions and Pass4sure dumps

Cisco's IP telephony providing has develop into so involved in such a short duration of time that to study it now involves a two-half, eight-day working towards course. Delegates can either attend both elements as a standalone choice (and purchase the direction exam if they select), or purchase a further 4 lessons and fade for the Cisco licensed Voice expert (CCVP) qualification.

global capabilities is an trustworthy practicing colleague of Cisco, and so closely follows its curriculum. It is among the greatest training providers for Cisco in the UK, having received the venerable largest provider, Azlan, in March 2006. as a result of the acquisition, it boasts a number of working towards centres across the South East, the Midlands and the North.

The direction focuses on CallManager (now referred to as Cisco Unified Communications supervisor), the name-processing application in Cisco's IP telephony providing. Delegates can pick to find out about CallManager edition 4, the most generally deployed windows-based mostly version, or version 5, which is Linux-based mostly.

ZDNet.co.uk reviewed the primary portion of the eight-day course and chose to focal point on version 4. the first portion lasts 5 days and covers about five-eighths of the content material. Delegates are given dates for the second part, which is three days in size, shortly after the conclusion of half one.

LocationWe reviewed this route at world abilities's London practicing centre, a industrious basement facility within the coronary heart of the metropolis's company district. international handicap also teaches the path in its Wakefield, Coventry and Wokingham working towards centres.

PrerequisitesGlobal expertise suggests that delegates abide already studied for each the Cisco certified Networking associate (CCNA) and Cisco Voice over IP (CVoice) certifications. however, truly, a few years of experience in each data and voice communications is probably going to exist satisfactory.

themes coveredCallManager working gadget, database and helping functions; CallManager deployment alternate options; establishing a name; Cisco telephony endpoints; CallManager configuration; configuring leavening switches to deal with voice; configuring Cisco IP Communicator tender cell; using the majority administration device; configuring gateways and trunks; configuring route plans; configuring hunt businesses; implementing class of manage; multi-site deployments; enabling features; configuring CallManager applications; including video; fine of provider; security; troubleshooting.

DeliveryThis route became delivered via an exceptionally a professional tutor who understood Cisco's IP telephony machine thoroughly. He became hardly unable to reply a question and he populated the course with factual titbits, a few of which abide been positive in purposeful terms ('bear in intelligence that passwords are reset when a CallManager update is utilized') and a few of which were just simple entertaining ('did you know why the MAC address incorporates that prefix?'). critically, he gave time-honored tips that could the category of questions that could seem in the examination.

however the tutor is completely employed via world expertise, he mentioned that he spends 20 % of his working time backyard the school scope with the purpose to keep up-to-date with IP and Cisco technology. His pretense seemed to exist substantiated, as he talked at length about industry routine and practical experiences.

The tutor additionally ran practical workouts configuring CallManager, which accounted for approximately 60 p.c of the course's duration. each and every step turned into certainly defined in precise aspect within the path notes, with the outcome that microscopic concept was required to observe the notes, arguably impeding the studying system.

We had only one criticism of the tutor: besides the fact that children he held the delegates' attention admirably, he did, during this reviewer's opinion, fail to entirely engage with the delegates in two-means communique.

quality of assisting materialsDelegates abide been provided with three hefty sets of notes overlaying the tutor's displays and purposeful exercises. These had been for essentially the most half provided by using Cisco, with small aboriginal additions. here is regular drill for a expertise working towards direction. Delegates had been additionally provided free ongoing aid by using e mail.

cost-addOur judgement on the value of the course for handicap delegates comes with one caveat. Delegates abide to believe the likelihood that they're going to necessity to deploy an IP telephony providing it really is made through Cisco. in terms of configuration, Cisco's IP telephony machine is significantly different to every other seller's machine, and so the course best has colossal expense for Cisco deployments. IT experts who are planning to set up non-Cisco IP telephony may accomplish more suitable in settling on one of the most many regularly occurring IP-telephony lessons on present from other providers.

youngsters, if the delegate is aspiring to deploy Cisco equipment, then the value-add from the eight days is considerable: a huge amount of hands-on competencies can also exist taken away.

Certification and further opportunitiesDelegates can pick to sit down an exam after the cessation of the course if they wish, and might then proceed to complete the CCVP in the event that they desire. The different lessons in the CCVP are every sole provided by world expertise. they are: pleasant of carrier; Cisco Voice over IP; Troubleshooting Cisco Unified Communications techniques; imposing Cisco Voice Gateways and Gatekeepers.

but a commemorate of warning: the total charges to attend every sole the classes crucial for the CCVP will set delegates returned a coloration over £10,000 — and that's the reason devoid of exam fees or VAT. The certification will also exist held for three years earlier than re-testing is required.

students might then pick to study for the plenty sought-after certified Cisco Internetwork knowledgeable (CCIE) qualification, Cisco's premiere certification.


Cisco and TDC convey greatest IP Telephony community in Pharmaceutical industry to Novo Nordisk | killexams.com actual Questions and Pass4sure dumps

COPENHAGEN, Denmark--(company WIRE)--April 19, 2005--Cisco methods (NASDAQ:CSCO) announced nowadays that Novo Nordisk A/S, a healthcare company and world leader in diabetes care, is upgrading its company network and deploying Cisco's information superhighway Protocol (IP) telephony solutions to assist converged voice and information traffic every sole the way through its operations in Denmark. The Cisco solution, which is designed with the goal of producing a satisfactory return on investment inside 2 years, is scheduled to exist thoroughly deployed with 12,000 IP telephones and 25,000 network ports within eight months. Novo Nordisk's community convergence undertaking will exist the largest IP telephony device in the Nordic locality as well because the biggest within the pharmaceutical trade international.

"For Novo Nordisk, relocating to Cisco IP Telephony changed into not a query of 'if' however 'when'," famed Sune Andersen, vice chairman, NNIT. "the size of impregnate tag downs is expected within the hundreds of thousands and, by working with Cisco and TDC, we've the means to compose a speedily migration to IP telephony."

Cisco Gold certified reseller partner, TDC/NetDesign, is working with NNIT, IT company to Novo Nordisk, to improve the enterprise's network and migrate the legacy cell device to Cisco's IP Telephony system.

New cell facets, similar to extension mobility, corporate directory shopping and Netwise's presence and availability administration integrated with Microsoft Outlook calendar and email applications, are being brought to enhance the supply, mobility and flexibility of voice communications and in the reduction of using expensive mobile phones.

The network improve will extend traffic skill via up to 10 times the use of Cisco Gigabit Ethernet technology and NNIT is planning to insert better tiers of safety with Cisco Self-Defending network know-how together with network Admission control.

"organizations every sole the way through Europe are realizing there's a sturdy enterprise case for converging voice and records," mentioned Tim Stone, Head of IP Communications in Europe, heart East & Africa (EMEA), Cisco systems. "Novo Nordisk's convergence assignment, the world's largest within the pharmaceutical business, is precise proof of the size of tag downs this is being finished with IP telephony these days."

Novo Nordisk's community improve will deliver in-line energy over Ethernet and contains the deployment of Cisco leavening 3550 sequence and leavening 3750 sequence switches, and 10 Gigabit supervisor module upgrades to existing leavening switches within the core network. The Cisco IP telephony solution includes a variety of Cisco 7900 sequence IP telephones and Cisco IP Communicator softphones with Cisco CallManager call processing software integrated with a 3rd birthday celebration contact management solution from Cisco AVVID associate, Netwise.

About Novo Nordisk

Novo Nordisk is a healthcare trade and an international leader in diabetes care. The company has the broadest diabetes product portfolio within the trade, together with probably the most superior items in the locality of insulin birth programs. furthermore, Novo Nordisk has a number one locality within areas similar to haemostasis administration, boom hormone therapy and hormone replacement therapy. Novo Nordisk manufactures and markets pharmaceutical products and capabilities that compose a significant distinction to patients, the clinical profession and society. With headquarters in Denmark, Novo Nordisk employs approximately 20,250 full-time employees in 78 international locations, and markets its items in 179 countries. Novo Nordisk's B shares are listed on the inventory exchanges in Copenhagen and London. Its ADRs are listed on the ny inventory change below the image 'NVO'. For greater tips, visit novonordisk.com.

About NNIT

NNIT is one among Denmark's main consultancies in the development, implementation and operation of IT solutions. For greater than 10 years, NNIT has applied new expertise and advanced methodologies to compose utility development, enterprise procedures and communication greater productive and beneficial. With its roots in the pharmaceutical trade, and as a company owned with the aid of Novo Nordisk, NNIT offers options that agree to the market's highest attribute and standardisation necessities.

nowadays, NNIT has more than 810 employees and a turnover in 2004 of DKK 800 million.

About TDC A/S

TDC trade solutions, a division within TDC A/S, strengthen and sell communications options and products to Danish companies and govt associations. TDC trade options is the market chief in Denmark and acquired remaining year the enterprise NetDesign which specialises in IP telephony and networking.

About Cisco techniques

Cisco systems, Inc. (NASDAQ:CSCO), the international leader in networking for the cyber web, this 12 months celebrates twenty years of dedication to expertise innovation, trade management and company sociable accountability. counsel on Cisco can also exist discovered at http://www.cisco.com. For ongoing information, gladden fade to http://newsroom.cisco.com. Cisco equipment in Europe is provided via Cisco techniques international BV, a unconditionally owned subsidiary of Cisco techniques, Inc.

Cisco, Cisco methods, and the Cisco techniques emblem are registered trademarks or emblems of Cisco systems, Inc. and/or its affiliates in the u.s. and certain other countries. every sole other trademarks outlined during this document are the property of their respective house owners. using the notice associate does not imply a partnership relationship between Cisco and another business. This document is Cisco Public counsel.


New Cisco Cert ambitions rich Media | killexams.com actual Questions and Pass4sure dumps

Citing experiences claiming billion dollar extend in on-demand media, Cisco has introduced a brand new certification for wealthy media in IP networks.

in keeping with the business, its new certification, wealthy Media Communications professional, will compose confident IT experts "have the practicing to improve the potential required to design, installation and manage wealthy media purposes to exploit new market calls for, together with integrating video conferencing functions privilege into a voice and facts network."

Cisco items the certification encompasses will embrace Cisco CallManager for IP telephony, Cisco harmony voice messaging and Cisco IP Contact facilities.

earning a wealthy Media expert certification will require keeping an lively Cisco licensed network associate (CCNA) certification and passing two additional checks:

  • 642-481 CRMC -- implementing Cisco IP Video convention rich Media Communications (ICRMC)
  • 642-432 CVOICE -- Cisco Voice Over IP (CVOICE)
  • The trade is providing the enforcing Cisco IP Video conference wealthy Media Communications (ICRMC) direction, a self-paced kind it expects most college students will comprehensive in about eight hours. The business's advised working towards for CVOICE is accessible as a 5-day instructor-led class, and as a number of e-discovering alternatives.


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    Implementing Cisco IP Telephony and Video, portion 1 (CIPTV1)

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    Book Extract: Implementing Cisco Unified Communications | killexams.com actual questions and Pass4sure dumps

    Chapter 1, Identifying Issues in a Multisite Deployment

    Deploying Cisco Unified Communications Manager between multiple sites requires an confiscate dial plan, confiscate bandwidth, a attribute of service (QoS) implementation and a design that can survive IP WAN failures.

    This chapter will warrant the issues that can arise in a multisite Cisco UC Manager deployment, and recommend specific solutions.

    Identifying Issues in a Multisite DeploymentTable of contents:1. Multisite deployment challenge overview2. attribute challenges3. Bandwidth challenges4. Availability challenges5. Dial contrivance challenges6. Overlapping and nonconsecutive numbers7. Fixed versus variable-length numbering plans8. Variable-length numbering, E.164 addressing and DID9. Optimized call routing and PSTN backup10. NAT and security issues

    Multisite deployment challenge overview

    Formerly known as Cisco call Manager (CCM), Cisco Unified Communications Manager (CUCM) multi-site deployment challenges can embrace the following:

  • Quality issues: Real-time communications of voice and video must exist prioritized over a packet-switching network. every sole traffic is treated equally by default in routers and switches. Voice and video are delay-sensitive packets that necessity to exist given priority to avoid laggard and jitter (variable delay), which would result in decreased voice quality.
  • Bandwidth issues: Cisco Unified Communications (Cisco UC) can embrace voice and video streams, signaling traffic, management traffic, and application traffic such as rich media conferencing. The additional bandwidth that is required when deploying a Cisco Unified Communications solution has to exist calculated and provisioned for to ensure that data applications and Cisco Unified Communications applications accomplish not overload the available bandwidth. Bandwidth reservations can exist made to applications through QoS deployment.
  • Availability issues: When deploying Cisco Unified Communications Manager (CUCM) with centralized call processing, IP Phones register with CUCM over the IP LAN and potentially over the WAN. If gateways in remote sites are using Media Gateway Control Protocol (MGCP) as a signaling protocol, they also depend on the availability of CUCM acting as an MGCP call agent. It is significant to implement fallback solutions for IP Phones and gateways in scenarios in which the connection to the CUCM servers is broken because of IP WAN failure. Fallback solutions also apply to H.323 gateways but are already created with H.323 dial peers in a proper H.323 gateway configuration.
  • Dial contrivance issues: Directory numbers (DN) can overlap across multiple sites. Overlapping dial plans and nonconsecutive numbers can exist solved by designing a robust multisite dial plan. Avoid overlapping numbers across sites whenever feasible for an easier design.
  • NAT and security issues: The use of private IP addresses within an enterprise IP network is very common. Internet Telephony Service Providers (ITSP) require unique public IP addresses to route IP Phone calls. The private IP addresses within the enterprise abide to exist translated into public IP addresses. Public IP addresses compose the IP Phones visible from the Internet and therefore matter to attacks.
  • Quality challenges

    IP networks were not originally designed to carry real-time traffic; instead, they were designed for resiliency and failing tolerance. Each packet is processed separately in an IP network, sometimes causing different packets in a communications stream to purchase different paths to the destination. The different paths in the network may abide a different amount of packet loss, delay, and laggard variation (jitter) because of bandwidth, distance, and congestion differences. The destination must exist able to receive packets out of order and resequence these packets. This challenge is solved by the use of Real-Time Transport Protocol (RTP) sequence numbers and traffic resequencing. When possible, it is best to not faith solely on these RTP mechanisms. Proper network design, using Cisco router Cisco Express Forwarding (CEF) switch cache technology, performs per-destination load sharing by default. Per-destination load sharing is not a faultless load-balancing paradigm, but it ensures that each IP current (voice call) takes the selfsame path.

    Bandwidth is shared by multiple users and applications, whereas the amount of bandwidth required for an individual IP current varies significantly during short lapses of time. Most data applications are very bursty, whereas Cisco real-time audio communications with RTP use the selfsame continuous-bandwidth stream. The bandwidth available for any application, including CUCM and voice-bearer traffic, is unpredictable. During peak periods, packets necessity to exist buffered in queues waiting to exist processed because of network congestion. Queuing is a term that anyone who has ever experienced air flight is chummy with. When you arrive at the airport, you must net in a line (queue), because the number of ticket agents (bandwidth) available to check you in is less than the current of traffic arriving at the ticket counters (incoming IP traffic). If congestion occurs for too long, the queue (packet buffers) gets filled up, and passengers are annoyed (packets are dropped). Higher queuing delays and packet drops are more likely on highly loaded, slow-speed links such as WAN links used between sites in a multisite environment. attribute challenges are common on these types of links, and you necessity to exploit them by implementing QoS. Without the use of QoS, voice packets experience delay, jitter, and packet loss, impacting voice quality. It is captious to properly configure Cisco QoS mechanisms cessation to cessation throughout the network for proper audio and video performance.

    This chapter was excerpted with permissions from Implementing Cisco Unified Communications Manager, portion 2, by Chris Olsen, Copyright 2008. For more information about this engage and other titles, gladden visit Cisco Press.

    During peak periods, packets cannot exist sent immediately because of interface congestion. Instead, the packets are temporarily stored in a queue, waiting to exist processed. The amount of time the packet waits in the queue, called the queuing delay, can vary greatly based on network conditions and traffic arrival rates. If the queue is full, newly received packets cannot exist buffered anymore and net dropped (tail drop). device 1-1 illustrates tail drop. Packets are processed on a first in, first out (FIFO) model in the hardware queue of every sole router interfaces. Voice conversations are predictable and constant (sampling is every 20 milliseconds by default), but data applications are bursty and greedy. Voice therefore is matter to degradation of attribute because of delay, jitter, and packet loss.

    Figure 1-1

    Click for a larger image

    Bandwidth challenges

    Each site in a multisite deployment usually is interconnected by an IP WAN, or occasionally by a metropolitan-area network (MAN) such as Metro Ethernet. Bandwidth on WAN links is limited and relatively expensive. The goal is to use the available bandwidth as efficiently as possible. Unnecessary traffic should exist removed from the IP WAN links through content filtering, firewalls, and access control lists (ACL). IP WAN acceleration methods for bandwidth optimization should exist considered as well. Any period of congestion could result in service degradation unless QoS is deployed throughout the network.

    Voice streams are constant and predictable for Cisco audio packets. Typically, the G.729 codec is used across the WAN to best use bandwidth. As a comparison, the G.711 audio codec requires 64 kbps, whereas packetizing the G.711 voice sample in an IP/UDP/RTP header every 20 ms requires 16 kbps plus the Layer 2 header overhead.

    Voice is sampled every 20 ms, resulting in 50 packets per second (pps). The IP header is 20 bytes, whereas the UDP header is 8 bytes, and the RTP header is 12 bytes. The 40 bytes of header information must exist converted to bits to device out the packet rate of the overhead. Because a byte has 8 bits, 40 bytes * 8 bits in a byte = 320 bits. The 320 bits are sent 50 times per second based on the 20-ms rate (1 millisecond is 1/1000 of a second, and 20/1000 = .02). So:

    .02 * 50 = 1 second320 bits * 50 = 16,000 bits/sec, or 16 kbps

    Voice packets are benign compared to the bandwidth consumed by data applications. Data applications can fill the entire maximum transmission unit (MTU) of an Ethernet frame (1518 bytes or 9216 bytes if jumbo Ethernet frames abide been enabled). In comparison to data application packets, voice packets are very small (60 bytes for G.729 and 200 bytes for G.711 with the default 20-ms sampling rate).

    In device 1-2, a conference bridge has been deployed at the main site. No conference bridge exists at the remote site. If three IP Phones at a remote site combine a conference, their RTP streams are sent across the WAN to the conference bridge. The conference bridge, whether using software or hardware resources, mixes the received audio streams and then sends back three unique unicast audio streams to the IP Phones over the IP WAN. The conference bridge removes the receiver's voice from his or her unique RTP stream so that the user does not experience reverberate because of the laggard of traversing the WAN link and mixing RTP audio streams in the conference bridge.

    Figure 1-2

    Click for a larger image

    Centralized conference resources cause bandwidth, delay, and capacity challenges in the voice network. Each G.711 RTP stream requires 80 kbps (plus the Layer 2 overhead), resulting in 240 kbps of IP WAN bandwidth consumption by this voice conference. If the conference bridge were not located on the other side of the IP WAN, this traffic would not necessity to traverse the WAN link, resulting in less laggard and bandwidth consumption. If the remote site had a CUCM region configuration that resulted in calls with the G.729 codec back to the main site, the software conferencing resources of CUCM would not exist able to mix the audio conversations. Hardware conferencing or hardware transcoder media resources in a voice gateway are required to accommodate G.729 audio conferencing. Local hardware conference resources would remove this need. every sole centrally located media resources (Music On Hold [MOH], annunciator, conference bridges, videoconferencing, and media termination points) suffer similar bandwidth, delay, and resource exhaustion challenges.

    Availability challenges

    When deploying CUCM in multisite environments, centralized CUCM-based services are accessed over the IP WAN. Affected services embrace the following:

  • Signaling in CUCM multisite deployments with centralized call processing: Remote Cisco IP Phones register with a centralized CUCM server. Remote MGCP gateways are controlled by a centralized CUCM server that acts as an MGCP call agent.
  • Signaling in CUCM multisite deployments with distributed call processing: In such environments, sites are connected via H.323 (non-gatekeeper-controlled, gatekeeper-controlled, or H.225) or Session Initiation Protocol (SIP) trunks.
  • Media exchange: RTP streams between endpoints located at different sites.
  • Other services: These embrace Cisco IP Phone Extensible Markup Language (XML) services and access to applications such as attendant console, CUCM Assistant, and others.
  • Figure 1-3 shows a Unified Communications network in which the main site is connected to a remote site through a centralized call-processing environment. The main site is also connected to a remote cluster through an intercluster trunk (ICT) representing a distributed call processing environment. The combination of both centralized and distributed call processing represents a hybrid call-processing model in which small sites use the CUCM resources of the main site, but big remote offices abide their own CUCM cluster. On the bottom left of device 1-3 is a SIP trunk, typically over a Metro Ethernet connection to an Internet Telephony Service Provider (ITSP). The capitalize of the SIP trunk is that the ITSP provides the gateways to the PSTN instead of your providing gateways at the main site.

    Figure 1-3

    Click for a larger image

    An IP WAN outage in device 1-3 will cause an outage of call-processing services for the remote site connected in a centralized fashion. The remote cluster will not suffer a call-processing outage, but the remote cluster will not exist able to dial the main site over the IP WAN during the outage. Mission-critical voice applications (voice mail, interactive voice response [IVR], and so on) located at the main site will exist unavailable to any of the other sites during the WAN outage.

    If the ITSP is using the selfsame links that allow IP WAN connectivity, every sole calls to and from the public switched telephone network (PSTN) will also exist unavailable.

    A deployment fancy the one shown in device 1-3 is considered badly designed because of the lack of IP WAN and PSTN backup.

    Dial contrivance challenges

    In a multisite deployment, with a sole or multiple CUCM clusters, dial contrivance design requires the consideration of several issues that accomplish not exist in single-site deployments:

  • Overlapping numbers: Users located at different sites can abide the selfsame directory numbers assigned. Because directory numbers usually are unique only within a site, a multisite deployment requires a solution for overlapping numbers.
  • Nonconsecutive numbers: Contiguous ranges of numbers are significant to summarize call-routing information, analogous to contiguous IP address ranges for route summarization. Such blocks can exist represented by one or a few entries in a call-routing table, such as route patterns, dial peer destination patterns, and voice translation rules, which keep the routing table short and simple. If each endpoint requires its own entry in the call-routing table, the table gets too big, lots of remembrance is required, and lookups purchase more time. Therefore, nonconsecutive numbers at any site are not optimal for efficient call routing.
  • Variable-length numbering: Some countries, such as the U.S. and Canada, abide fixed-length numbering plans for PSTN numbers. Others, such as Mexico and England, abide variable-length numbering plans. A problem with variable-length numbers is that the complete length of the number dialed can exist determined only by the CUCM route contrivance by waiting for the interdigit timeout. Waiting for the interdigit timeout, known as the T.302 timer, adds to the post-dial delay, which may rile users.
  • Direct inward dialing (DID) ranges and E.164 addressing: When considering integration with the PSTN, internally used directory numbers abide to exist related to external PSTN numbers (E.164 addressing). Depending on the numbering contrivance (fixed or variable) and services provided by the PSTN, the following solutions are common:
  • Each internal directory number relates to a fixed-length PSTN number: In this case, each internal directory number has its own dedicated PSTN number. The directory number can, but does not abide to, match the least-significant digits of the PSTN number. In countries with a fixed numbering plan, such as the North American Numbering contrivance (NANP), this usually means that the four-digit office codes are used as internal directory numbers. If these are not unique, digits of office codes or administratively assigned site codes might exist added, resulting in five or more digits being used for internal directory numbers.

    Another solution is to not reuse any digits of the PSTN number but to simply map each internally used directory number to any PSTN number assigned to the company. In this case, the internal and external numbers accomplish not abide anything in common. If the internally used directory number matches the least-significant digits of its corresponding PSTN number, significant digits can exist set at the gateway or trunk. Also, common external phone number masks, transformation masks, or prefixes can exist configured. This is steady because every sole internal directory numbers are changed to fully qualified PSTN numbers in the selfsame way. Another instance is if the internal directory number is composed of parts of the PSTN number and administratively assigned digits such as site codes plus PSTN station codes, or different ranges, such as PSTN station codes 4100 to 4180 that map to directory numbers 1100 to 1180, or totally independent mappings of internal directory numbers to PSTN numbers. In that case, one or more translation rules abide to exist used for incoming calls, and one or more calling party transformation rules, transformation masks, external phone number masks, or prefixes abide to exist configured.

  • No DID champion in fixed-length numbering plans: To avoid the requirement of one PSTN number per internal directory number when using a fixed-length numbering plan, it is common to disallow DID to an extension. Instead, the PSTN trunk has a sole number, and every sole PSTN calls routed to that number are sent to an attendant, auto-attendant, receptionist, or secretary. From there, the calls are transferred to the confiscate internal extension.
  • Internal directory numbers are portion of a variable-length number: In countries with variable-length numbering plans, a typically shorter "subscriber" number is assigned to the PSTN trunk, but the PSTN routes every sole calls starting with this number to the trunk. The caller can add digits to identify the extension. There is no fixed number of additional digits or total digits. However, there is a maximum, usually 32 digits, which provides the license to select the length of directory numbers. This maximum length can exist less. For example, in E.164 the maximum number is 15 digits, not including the country code. A caller simply adds the confiscate extension to the company's (short) PSTN number when placing a call to a specific user. If only the short PSTN number without an extension is dialed, the call is routed to an attendant within the company. Residential PSTN numbers are usually longer and accomplish not allow additional digits to exist added; the feature just described is available only on trunks.
  • Type of Number (TON) in ISDN: The calling number (the Automatic Number Identification [ANI]) of calls being received from the PSTN can exist represented in different ways:

    - As a seven-digit subscriber number- As a ten-digit number, including the locality code- In international format with the country code in front of the locality code

    To standardize the ANI for every sole calls, the format that is used must exist known, and the number has to exist transformed accordingly.

  • Optimized call routing: Having an IP WAN between sites with PSTN access at every sole sites allows PSTN toll bypass by sending calls between sites over the IP WAN instead of using the PSTN. In such scenarios, the PSTN should exist used as a backup path only in case of WAN failure. Another solution, which extends the idea of toll bypass and can potentially reduce toll charges, is to also use the IP WAN for PSTN calls. With tail-end hop-off (TEHO), the IP WAN is used as much as possible, and the gateway that is closest to the dialed PSTN destination is used for the PSTN breakout.
  • Overlapping and nonconsecutive numbers

    In device 1-4, Cisco IP Phones at the main site use directory numbers 1001 to 1099, 2000 to 2157, and 2365 to 2999. At the remote site, 1001 to 1099 and 2158 to 2364 are used. These directory numbers abide two issues. First, 1001 to 1099 overlap; these directory numbers exist at both sites, so they are not unique throughout the complete deployment. This causes a problem: If a user in the remote site dialed only the four digits 1001, which phone would ring? This issue of overlapping dial plans needs to exist addressed by digit manipulation. In addition, the nonconsecutive use of the scope 2000 to 2999 (with some duplicate numbers at the two sites) would require a significant number of additional entries in call-routing tables because the ranges can hardly exist summarized by one (or a few) entries.

    Figure 1-4

    Click for a larger image

    Fixed versus variable-length numbering plans

    A fixed numbering contrivance features fixed-length locality codes and local numbers. An open numbering contrivance features variance in length of locality code or local number, or both, within the country.

    Table 1-1 contrasts the NANP and a variable-length numbering contrivance -- Germany's numbering contrivance in this example.

    Table 1-1

    Component Description Fixed Numbering contrivance (NANP) Variable-Length Numbering contrivance (Germany) Country code A code of one to three digits is used to gain the particular telephone system for each nation or special service. Obtain the E.164 standard from http://itu.org to observe every sole international country codes. 1 49 Area code Used within many nations to route calls to a particular city, region, or special service. Depending on the nation or region, it may also exist called a numbering contrivance area, subscriber trunk dialing code, national destination code, or routing code. Three digits Three to five digits Subscriber number Represents the specific telephone number to exist dialed, but it does not embrace the country code, locality code (if applicable), international prefix, or trunk prefix. Three-digit exchange code plus a four-digit station code Three or more digits Trunk prefix The initial digits to exist dialed in a domestic call, before the locality code and the subscriber number. 1 0 Access code A number that is traditionally dialed first "to net out to the PSTN," used in PBXs and VoIP systems. 9 0 International prefix The code dialed before an international number (country code, locality code if any, and then subscriber number). 011 00 or + (+ is used by cell phones)

    Examples:

  • Within the U.S.: 9-1-408-555-1234 or 1-555-1234 (within the selfsame locality code)
  • U.S. to Germany: 9-011-49-404-132670
  • Within Germany: 0-0-404-132670 or 0-132670 (within the selfsame locality code)
  • Germany to the U.S.: 0-00-1-408-555-1234 (Note: the 1 in 00-1-408 is the U.S. country code, not the trunk prefix.)
  • The NANP PSTN number is 408-555-1234, DID is not used, and every sole calls placed to the main site are handled by an attendant. There is a remote site in Germany with the E.164 PSTN number +49 404 13267. Four-digit extensions are used at the German location, and DID is allowed because digits can exist added to the PSTN number. When calling the German office attendant (not knowing a specific extension), U.S. users would dial 9-011-49-404-13267. Note how the + is replaced by the international prefix 011 and the access code 9. If the phone with extension 1001 should exist called directly, 9-011-49-404-13267-1001 has to exist dialed.

    Variable-length numbering, E.164 addressing and DID

    Figure 1-5 illustrates an instance in which the main site with CUCM resides in the U.S. and a remote site without CUCM resides in Germany. The NANP PSTN number in the U.S. is 408-555-1234. Note that DID is not used, because every sole calls placed to the main site are handled by an attendant. A remote site in Germany has PSTN number +49 404 13267. Four-digit extensions are used at the German location, and DID is allowed because digits can exist added to the PSTN number. When calling the German office attendant (not knowing a specific extension), U.S. users would dial 9-011-49-404-13267. If the phone with extension 1001 should exist called directly, 9-011-49-404-13267-1001 has to exist dialed.

    Figure 1-5

    Click for a larger image

    The logic of routing calls by CUCM over the WAN or through the PSTN is appropriately transparent to the phone user.

    Optimized call routing and PSTN backup

    There are two ways to rescue costs for PSTN calls in a multisite deployment:

  • Total bypass: Calls between sites within an organization that use the IP WAN instead of the PSTN. The PSTN is used for intersite calls only if calls over the IP WAN are not possible—either because of a WAN failure or because the call is not admitted by call Admission Control (CAC).
  • Tail-end hop-off (TEHO): Extends the concept of toll bypass by also using the IP WAN for calls to the remote destinations in the PSTN. With TEHO, the IP WAN is used as much as possible, and PSTN breakout occurs at the gateway that is located closest to the dialed PSTN destination. Local PSTN breakout is used as a backup in case of IP WAN or CAC.
  • In the instance shown in device 1-6, a call from Chicago to San Jose would exist routed as follows:

  • The Chicago CUCM Express user dials 9-1-408-555-6666, a PSTN phone located in San Jose.
  • The call is routed from Chicago CUCM Express Router to the San Jose CUCM cluster over the IP WAN with either SIP or H.323.
  • The San Jose CUCM routes the call to the San Jose gateway, which breaks out to the PSTN with what now becomes a local inexpensive call to the San Jose PSTN.
  • The San Jose PSTN Central Office routes the call, and the phone rings.
  • Figure 1-6

    Click for a larger image

    If the WAN were unavailable for any reason before the call, the Chicago Gateway would abide to exist properly configured to route the call with the confiscate digit manipulation through the PSTN at a potentially higher toll cost to the San Jose PSTN phone.

    NAT and security issues

    In single-site deployments, CUCM servers and IP Phones usually use private IP addresses because there is no necessity to communicate with the outside IP world. NAT is not configured for the phone subnets, and attacks from the outside are impossible. In multisite deployments, however, IP Security (IPsec) virtual private network (VPN) tunnels can exist used between sites. The VPN tunnels allow only intersite communication; access to the protected internal networks is not feasible from the outside—only from the other site through the tunnel. Therefore, attacks from the outside are blocked at the gateway. To configure IPsec VPNs, the VPN tunnel must exist configured to terminate on the two gateways in the different sites. Sometimes this is not possible; for instance, the two sites may exist under different administration, or perhaps security policies accomplish not allow the configuration of IPsec VPNs.

    In such a case, or when connecting to a public service such as an ITSP, NAT has to exist configured for CUCM servers and IP Phones. Cisco calls this Hosted NAT Traversal for Session verge Controllers.

    In device 1-7, Company A and Company B both use IP network 10.0.0.0/8 internally. To communicate over the Internet, the private addresses are translated into public IP addresses. Company A uses public IP network A, and Company B uses public IP network B. every sole CUCM servers and IP Phones can exist reached from the Internet and communicate with each other.

    As soon as CUCM servers and IP Phones can exist reached with public IP addresses, they are matter to attacks from the outside world, introducing potential security issues.

    Figure 1-7

    Click for a larger image

    This download first appeared at searchunifiedcommunications.com

    Windstream Announces Nationwide Availability of Its Enterprise-Class Unified Communications as a Service | killexams.com actual questions and Pass4sure dumps

    LITTLE ROCK, Ark., Jul 07, 2014 (GLOBE NEWSWIRE via COMTEX) --

    Windstream WIN, +1.25% a leading provider of advanced network communications, today announced the nationwide availability of its enterprise-class Unified Communications as a Service (UCaaS). Windstream's UCaaS is a fully managed cloud-based solution, offered to enterprise and mid-sized businesses for a flat monthly rate. The UCaaS product suite is hosted in Windstream's secure data centers and leverages Windstream's high-quality Multiprotocol Label Switching (MPLS) networking with attribute of Service (QoS) for optimized performance.

    "Our UC solution helps businesses create a more collaborative and connected workforce, providing anytime, anywhere access to a scope of UC options, using a single, consistent interface," said Matt Preschern, senior vice president and enterprise chief marketing officer at Windstream. "Our trade customers are seeking solutions that tender trade continuity, security and increased productivity – UCaaS, supported by their team of experts and entirely hosted and managed by Windstream, from their data heart to the MPLS network, provides the smart solutions and personalized service their customers necessity to wait on their businesses succeed."

    According to industry research firm, Frost & Sullivan, the hosted IP telephony and UCC market in North America, which includes UCaaS, is projected to gain $13.23 billion in total market size by 2020, growing at a compound annual growth rate of 28.4%.1 Developed in 2013, Windstream's UCaaS solutions achieved success during beta testing and abide been designed to meet the evolving technology needs of enterprise and mid-sized businesses. UCaaS seamlessly integrates with existing customer tools, such as call centers, Exchange products and sociable media platforms, and its sole interface enables a higher smooth of application integration and customization. Bundled solutions tender easy, affordable implementation of Unified Communications, and embrace services such as analog or VoIP lines, unified messaging, desktop video and a variety of other mobility solutions.

    Windstream's MPLS network provides private and secure networking for speedily and safe communications between multiple sites and locations, as well as the scalability required to adjust as trade needs change, as well as QoS ensuring elevated attribute voice communications. Windstream's UCaaS solutions feature 24x7 remote and onsite monitoring and support. Windstream's cloud computing, data storage and managed services operate out of its nationwide network of enterprise-class data centers.

    "The introduction of seamless and uniform real-time communication platforms is driving enterprise adoption of unified communications solutions," said Bill Haskins, senior analyst and colleague at Wainhouse Research. "Windstream UCaaS is a cloud-based UC solution that yields a number of benefits and advantages, including the aptitude to reduce or eliminate capital investment, reduce management complexity and enhance the flexibility of upgrades; plus, it's a multi-vendor solution that can exist tailored to meet an enterprise's specific requirements."

    As portion of its UCaaS solution, Windstream has forged partnerships with a number of market leaders, including Avaya, Cisco, Mitel and ShoreTel. For more information about Windstream's enterprise class data, voice, network and cloud solutions, gladden visit www.windstreambusiness.com.

    1 Includes access revenues. Frost & Sullivan. July 2014

    About Windstream

    Windstream WIN, +1.25% a FORTUNE 500 and S&P 500 company, is a leading provider of advanced network communications, including cloud computing and managed services, to businesses nationwide. The company also offers broadband, phone and digital TV services to consumers primarily in bucolic areas. For more information, visit www.windstream.com.

    CONTACT: Windstream Contact: Tanja Jameson 501.748.7236 tanja.jameson@windstream.com

    Copyright (C) 2014 GlobeNewswire, Inc. every sole rights reserved.


    Cisco Announces ‘Networkers Innovation Awards 2011’ Winners | killexams.com actual questions and Pass4sure dumps

    WEBWIRE – Friday, September 30, 2011

    Awards recognize companies at the forefront of implementing and applying innovative technologies

    LIMA, Peru – Cisco announced today the winners of the Innovation Awards Peru 2011 in the categories of Data Center, Collaboration (unified communications and video) and IP NGN (Internet Protocol Next-Generation Network). Announced at Cisco´s annual Networkers event,these awards recognize the achievements of companies in the northern portion of South America, Central America and the Caribbean that adopt advanced technologies.

    The judges, eight Cisco representatives, evaluated solutions and innovative projects that clearly showed the benefits of implementation, a measurable and significant return of investment, improvements in operating and administrative processes, and new services enabled by the efficient use of Cisco® technologies.

    “Innovation Awards” Networkers Peru 2011 Winners

    • Data heart Category : Orbis Access/ Bermuda

    Orbis, a private investment fund in Bermuda, launched a new “Go to Market” (GTM) model called Orbis Access, which has grandiose potential to generate private investment and experience for the company. Orbis needed a cloud solution that, from the very first day of execution, was able to champion approximately 1 million users. The public cloud solution includes Cisco Nexus® servers together with the Cisco Unified Computing System™ B-Series products. The implementation of the Cisco Data heart will allow Orbis to scale vertically and horizontally, and the goal is to abide multiple Cisco data centers around the world.

    • Collaboration Category : Organo Electoral Plurinacional – Supreme Electoral Tribunal – Bolivia

    The OPE (Bolivian Electoral Organization) had to create security mechanisms and data storage for the departmental and municipal elections of 2010. For this purpose, it designed a new voice and data national network, involving the re-engineering of the existing LAN network at a national smooth to fill the security requirements with a replete equipment infrastructure, integrate a highly secure unified communications platform, facilitate communications at a national level, and integrate video systems, apart from deploying a wireless network.

    Cisco had the capacity to provide every sole the required equipment, with one standardized infrastructure in the main data heart as well as the data centers in the nine cities of the country, linking them in a highly secure and simple way through a unique and accessible way to manage every sole the points. The implemented solutions are in the areas of routing, switching, security, optimization, wireless and IP telephony. The OPE has become a very significant technological reference in Bolivia, setting the standard for public and private institutions for technological implementations.

    • IP-NGN Category- Entel S.A – Bolivia

    Entel, one of the most significant telecommunications companies in Bolivia, needed to implement a next-generation network at a national smooth as a universal platform to deliver convergened voice, data, video and mobile services to every sole market segments. It needed to exist able to bring customers added-value services “anytime, anywhere and from any device”. The project was driven by the company policy of supporting the initiatives of the Bolivian government of sociable inclusion and the improvement of the citizens’ attribute of life.

    As a result, Entel has improved substantially the attribute of service for its current customers, and the new network has started to champion the access of mobile users in 3.75G.

    The winning projects in each category were announced at the Innovation Awards ceremony during Cisco Networkers 2011 Peru. In attendance were Carlos Domínguez, Cisco senior vice president; and Alvaro Merino-Reyna, common manager, Cisco Peru.

    About CiscoCisco (NASDAQ: CSCO) is the worldwide leader in networking that transforms how people connect, communicate and collaborate. Information about Cisco can exist found at http://www.cisco.com. For ongoing news, gladden fade to http://newsroom.cisco.com.

    Cisco and the Cisco logo are trademarks or registered trademarks of Cisco and/or its affiliates in the U.S. and other countries. A listing of Cisco’s trademarks can exist found at www.cisco.com/go/trademarks.Other third-party trademarks mentioned are the property of their respective owners. The use of the word colleague does not imply a partnership relationship between Cisco and any other company.

    About Networkers Innovation Awards

    Cisco Innovation Awards at Cisco Networkers are an significant initiative for the achievements of organizations adopting new and advanced technologies. The awards direct to recognize the companies at the forefront of implementing and applying innovative technologies. The judging of advanced technology initiatives is based on the benefits of the implementation, a significant improvement in processes, and innovative services through the efficient use of Cisco technologies. For further information, gladden visit: http://www.networkerscisco.com/web/PE/networkers2011/innovation/

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    References :


    Vimeo : https://vimeo.com//220776036
    Blogspot : http://killexamz.blogspot.com/2017/05/killexamscom-300-070-braindumps-and.html
    Youtube : https://youtu.be/ESrDs6ADGjw
    weSRCH : https://www.wesrch.com/business/prpdfBU1HWO000HDYA
    Dropmark : http://killexams.dropmark.com/367904/11477164
    Issu : https://issuu.com/trutrainers/docs/300-070
    Wordpress : http://wp.me/p7SJ6L-jN
    Scribd : https://www.scribd.com/document/357087777/Pass4sure-300-070-Practice-Tests-with-Real-Questions
    Dropmark-Text : http://killexams.dropmark.com/367904/12038555
    RSS Feed : http://feeds.feedburner.com/Real300-070QuestionsThatAppearedInTestToday
    publitas.com : https://view.publitas.com/trutrainers-inc/looking-for-300-070-exam-dumps-that-works-in-real-exam
    Google+ : https://plus.google.com/112153555852933435691/posts/2G9oTG6zCJd?hl=en
    Calameo : http://en.calameo.com/account/book#
    Box.net : https://app.box.com/s/4eiep4ldrsza4editirzeyi597jnxm3s
    zoho.com : https://docs.zoho.com/file/4b1e162e6f446d97941ba92086610dc0a2cac






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