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920-316 SCCS/Express/TAPI/Web Client/Agent

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920-316 exam Dumps Source : SCCS/Express/TAPI/Web Client/Agent

Test Code : 920-316
Test appellation : SCCS/Express/TAPI/Web Client/Agent
Vendor appellation : Nortel
: 70 actual Questions

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Nortel SCCS/Express/TAPI/Web Client/Agent

Certification Watch: New Solaris security exam Enters Beta | actual Questions and Pass4sure dumps

download the authoritative guide: Cloud Computing 2019: the spend of the Cloud for competitive competencies

the brand new sun licensed Solaris protection Administrator examination is now purchasable in beta format. The beta length will remaining unless March eight.

examination goals should be despatched to these you register for the beta at solar's web site.

Microsoft presents Free .web exam to latest MCSDs

starting March 1, MCSDs who licensed on the visible Studio 6.0 track can accept a free voucher remarkable for examination 70-300 from the MCSD for .internet song. exam 70-300: inspecting requirements and Defining .web solution Architectures is one in every of four core checks required for MCSD for .internet certification. It always prices $a hundred twenty five. thus far, more than forty one,000 individuals beget earned the MCSD title. examination 70-300 went reside Feb. 10. The free voucher requisite to be used by Sept. 30. present MCSDs can apply for a voucher in the course of the MCP snug web web page. For greater details on MCSD .web argue with GoCertify's MCSD .net page.;n=203;c=204660774;s=9478;x=7936;f=201812281339040;u=j;z=TIMESTAMP;a=20403972;e=i Marimba Certification software Reborn

Marimba's certification software has emerged from its redecorate with two certifications that IT authorities can earn. the first, licensed Marimba knowledgeable (CMP), requires passage of 2-four online checks, reckoning on your choice of music. options consist of laptop administration Administrator, Server administration Administrator, utility Packager, and gadget Architect.

The 2d even of Marimba certification, licensed Marimba consultant, requires that you simply first merit CMP certification, plus whisk a arms-on purposeful examination that lasts someday. For more assistance observe GoCertify's Marimba web page.

SAS foundation and superior Programming checks are animate March 29

The final models of the SAS foundation Programming and SAS superior Programming exams are scheduled to head live in their ultimate (non-beta) structure with the aid of March 29. in contrast to earlier SAS checks, these don't appear to be edition-particular. This reflects the brand new corporation of the SAS certification application, by which certifications are designed round job position levels. For more recommendation on vulgar obtainable SAS certifications observe their SAS page.

Nortel Networks Releases Two New Design professional exams

you can now merit Nortel Networks Design expert (NNCDS) certifications in VoIP CallPilot 2.0 and SCCS/categorical/TAPI/web client/Agent.

the new CallPilot 2.0 (#920-155) examination is one in every of three required to merit NNCDS - VoIP CallPilot 2.0 certification. Completion of the VoIP expertise (#920-801) and Meridian 1 Database (#920-318) checks is furthermore required to merit VoIP certification.that you could merit NNCDS - SCCS/express/TAPI/net client/Agent certification by using correctly completing the new SCCS/specific/TAPI/web customer/Agent (#920-316) exam. The exams are proctored by means of Prometric.

it truly is occupied with this edition of Certification Watch. delight maintain your certification information and assistance coming to

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    Asterisk With workplace Communications Server 2007 | actual Questions and Pass4sure dumps

    Chief expertise Officer and govt Editor

    originally posted on VoIP & gadgets blog, privilege here:

    Asterisk logoOCS 2007 logoAsterisk (news - Alert) integration with office Communications Server 2007 is a favored theme in the unified communications group. You would not feel integrating an open source Linux-primarily based reply (Asterisk) with Microsoft's proprietary OCS 2007 UC platform could be well-known, but it surely is. basically, I even beget my ear to both the Microsoft OCS 2007 group and the Asterisk group and the combination of these two worlds is something I hear the entire time. Some inside the Asterisk neighborhood poo-poo the credence of integrating with Microsoft OCS 2007. Now, no matter if or not it's on account of some anti-Microsoft zeal or that Asterisk "can finish anything - simply supply the open source group time to construct it" mentality, I ought to respond that OCS 2007 is here to dwell and is doing very well. i'm alert about a number of Fortune 1000 corporations, including security-mindful predominant banks, that beget deployed OCS 2007.

    happily, no longer vulgar within the Asterisk community beget such blind hatred for Microsoft that they are inclined to are trying and combine with Microsoft OCS 2007, arguably the best UC platform available on the market today. There are some points that Asterisk simply doesn't have, equivalent to a characteristic prosperous softphone client with video support. The office Communicator customer integrates presence, appellation handle, IM, collaboration, video, and extra. privilege here's a pattern of me using it.

    OCS Communicator video call

    Of path, which you could spend X-Lite, a free SIP client, with Asterisk, nonetheless it's functionality and integration is rarely as feature-rich as the total OCS platform. though if cost is your only issue, Microsoft's OCS licensing may be too a total lot for some.

    So in any adventure, what is the issue in integrating Asterisk with OCS 2007? well, as I even beget written in the past, the leading challenge is that Asterisk uses SIP over UDP and OCS 2007 requires SIP over TCP.

    There are some workarounds despite the fact. One is to installation an unsupported patch onto Asterisk that adds SIP over TCP guide. in my view, i might avert this notwithstanding, considering any future updates might negate the patch and probably raze the "legitimate" Asterisk. further, SIP over TCP isn't very classic (apart from in OCS 2007 environments). a more robust alternative is to install sipX, an open-supply SIP proxy which handles each SIP/TCP and SIP/UDP (news - Alert) and acts as a gateway between Asterisk and OCS 2007. that you could spend OpenSER as a SIP UDP to SIP TCP gateway as neatly. really it seems dote this:Asterisk < SIP/UDP > OpenSER < SIP/TCP > OCS Mediation Server < uaCSTA:TLS|TCP > OCS Server

    On a connected word, pbxnsip (information - Alert), a Linux-based IP-PBX platform presents fine & handy integration with OCS 2007. try the pbxnsip with OCS 2007 wiki. you can additionally spend the remarkable value pbxnsip outfit as an inexpensive SIP UDP to SIP TCP gateway in case you requisite and spend this along side Asterisk. definitely, pbxnsip will furthermore be with ease integrated OCS 2007 and with change 2007 UM (Unified Messaging). try my plenary evaluate of pbxnsip for some historical past on the product.

    a new undertaking http://sourceforge.web/initiatives/asterisk-dotnet (Asterisk LCS gateway) to OCS 2007 claims to present Asterisk to OCS integration, nonetheless it seems dote a drudgery in development.

    here's strictly from the rumor-mill, but I heard that SIP over UDP champion "might also" be coming in the next free up of OCS 2007 - rumored to be called OCS 2007 R2. that could negate the want for a TCP-to-UDP SIP gateway and allow for direct Asterisk to OCS integration. further, Asterisk 1.6 (nonetheless beta) has SIP TCP inbuilt - and a few beget efficiently gotten it to drudgery at once with OCS Mediation server. So in spite of the fact that Microsoft would not present SIP over UDP assist in the next release, it feels dote Asterisk is adding SIP over TCP support.

    I should noiseless point out that Asterisk lacks CSTA champion and alas OpenSER does not attend it either. while CSTA is never required for OCS 2007 to Asterisk integration, CSTA provides tighter integration and advanced points comparable to RCC (far off call handle), presence integration, and twin-forking with presence gateways. further, CSTA offers for 3rd party CTI purposes, together with predictive dialers, superior ACD, and IVRs that leverage CSTA for speaking with the PBX (information - Alert).

    One reply I discovered that adds rudimentary CSTA capabilities to Asterisk is offered with the aid of M-Networks. Their solution, known as the Unified appellation handle Gateway (information - Alert), services as an intermediary that enables the Microsoft are animate Communications Server (LCS 2005) and Asterisk PBX to talk. moreover remote call control, the application provides for laptop to cell calling that allows for clients to spend their desktop as a cell. while it become designed for LCS 2005, in thought the application could drudgery on its successor, OCS 2007. here's a few of its facets:

  • TCP <-> UDP SIP Proxy - permits workstation to cell calls from Communicator 2005 without the requisite for extra application (ie. OpenSER)
  • SIP URI normalization - Converts sip: or tel: URI's to Asterisk extensions. furthermore allows for for remedy caller identity presentation for laptop to interior extension calls.
  • e.164 attend - plenary attend for the e.164 numbering usual
  • even though, if I stand in intellect this product as it should be it runs silently within the inheritance as a windows provider processing CSTA requests from workplace Communicator to manage your Asterisk PBX. thus, each computing device would ought to install this software, which is a deal-breaker for me. i'd rather beget a server-primarily based utility or hardware solution.

    Office Communicator it be captious to factor out that the CSTA common provides for third party CTI applications together with superior ACDs and IVRs which makes spend of CSTA for speaking with a PBX. this is vital in stupendous appellation centers. Asterisk not featuring CSTA functionality definitely is never the dying knell for proposing integration with other software. Asterisk supports their AGI/quick AGI (Asterisk Gateway Interface) which enables builders to write applications.

    additional, the Asterisk supervisor API adds third-birthday party appellation handle and might be concept of as a CSTA/TSAPI equivalent. Some Asterisk-based mostly options present TAPI-primarily based computing device appellation control as smartly, but TAPI may not attend you with OCS 2007 integration - exceptionally with the OCS Communicator presence/VoIP client (shown to the right).

    Let's appear at the routing once more:Asterisk < SIP/UDP > OpenSER < SIP/TCP > OCS Mediation Server < uaCSTA:TLS|TCP > OCS Server

    I suppose instead of this convoluted approach constructing a CSTA-to-AMI (Asterisk manager Interface) could present more advantageous integration with Asterisk. since OpenSER is not dialog stateful, you can not forward the repute of the SIP purchasers to OCS crucial for presence and other performance. It cannot for illustration be alert of when a specific session has been terminated. Asterisk in spite of this is dialog stateful and is privy to vulgar classes constantly. With uaCSTA interaction between OCS 2007 and Asterisk the consumer-state and the VoIP-cell-state will furthermore be exchanged.The big query is will anybody in the Asterisk group write the code to finish CSTA-to-AMI? anyone?

    Of direction, if you beget to beget CSTA for the superior telephony features today, you might appear privilege into a SIP/CSTA gateway. There are a number of available on the market. UniGone TelServer offers one. Mitel developed one certainly with OCS 2007 in intellect however I consider it really works above vulgar with their IP-PBX - identical goes for Nortel. I believe SIP/CSTA options which are "IP-PBX agnostic" are furthermore obtainable from Audiocodes, Dialogic, and Quintum.

    I should noiseless aspect out that OCS truly makes spend of uaCSTA (person agent CSTA) which refers to transporting ECMA-323 (CSTA XML) messages over a SIP session. uaCSTA leverages SIP mechanisms to supply an extensible set of elements to attend purposes. uaCSTA can be applied through a yoke of different types of SIP user brokers, together with directly by way of a SIP user agent on a SIP cellphone. uaCSTA can furthermore be carried out by using a SIP B2BUA to augment third-celebration appellation handle performance or via a proxy server that's front-ending a PBX.

    I additionally came about throughout a Microsoft TechNet thread that pointed out, "I drudgery at an organization this is developed a SIP/CSTA gateway for Asterisk and MS OCS. or not it's a requisites compliant TR/87 gateway that uses the manager API in Asterisk and SIP to MS OCS. It can be used for far off call control (CTI integration), location based mostly forwarding and device monitoring." The poster is from, a company that prosperous Tehrani wrote about as well as different TMCnet writers. So here's an extra expertise SIP/CSTA answer. I despatched an e-mail to LiteScape for more info however am ready for his or her reply.

    Conclusion:Asterisk integration with OCS 2007 is going on nowadays, but you ought to soar through hoops to accept it to work. extra, in case you requisite superior points dote RCC and twin-forking, you'll requisite a SIP/uaCSTA gateway -- some options of which I up to now mentioned. SIP over UDP guide "may additionally" be coming in the subsequent liberate of OCS 2007 which might negate the want for a TCP-to-UDP SIP gateway and allow for direct Asterisk to OCS integration. also, Asterisk 1.6 beta has SIP TCP built in and a few beget efficaciously gotten it to drudgery at once with OCS Mediation server. you'll nevertheless want a SIP/uaCSTA gateway of some sort except some entrepreneurial Asterisk guru writes a CSTA-to-AMI add-on for Asterisk.

    finally, i am informed that pbxnsip could beget uaCSTA guide very shortly. besides the fact that children pbxnsip is rarely Asterisk, it's yet another open source IP-PBX reply and a extremely characteristic-rich one at that. it be very handy to combine with OCS 2007 quite simply through adding a SIP trunk. The pbxnsip wiki explains the steps very smartly.

    [Correction: I mentioned pbxnsip was open source. This was due to a conversation I had with pbxnsip about them 'potentially' making their solution open source. They are currently not open source, however, I should point out that it is developed in C++ and is therefore easily compiled to hasten on multiple operating systems, including Windows, Linux, etc.]

    here's a screenshot of the leading step performed by means of the pbxnsip web admin:

    OCS Mediation Trunk

    [ Back To's Homepage ]

    The next-generation customer contact focus | actual Questions and Pass4sure dumps

    graphic this: instead of having to operate an expensive appellation core tied to a actual location, you may beget created a digital, multimedia contact focus staffed by using brokers working from home or in far workplaces, linked through a voice-over-IP community. you might beget launched cash-saving, self-carrier consumer purposes based on standards corresponding to XML and VoiceXML that slit back the weight to your agents. you could beget built-in your CRM application with computer-mobile integration (CTI) in order that returned-end database assistance on purchasers is attainable to brokers within the profile of monitor pops.

    consumers attain your contact core any approach they requisite - voice name, electronic mail,chat and precise-time collaboration over the web. The client event is constant across vulgar media, and each interaction is immediately mentioned so updated information is obtainable privilege away. The contact middle blends supplies, and manages and studies on vulgar contacts via a gauge software suite.

    is that this imaginative and prescient hype or fact? The reply is a bit of each. This contact middle is feasible with latest technologies, but full-blown implementations are nevertheless few and far between. besides the fact that children, consultants are confident that VoIP-primarily based, multichannel contact facilities are the way forward for client members of the family, so or not it's captious to start planning.

    The next-era contact core requires two simple architectural shifts. within the community infrastructure you should shift from the TDM/PBX world to VoIP. And in the applications infrastructure you requisite to migrate call routing and reporting, traditionally handled through automatic call distributors (ACD), from closed switching systems to open, industry-typical servers.

    Open structure

    Open structure offers the vow of infrastructure-agnostic purposes. Routing, reporting and management capabilities can hasten on a server that talks to the switching infrastructure. Open functions can seamlessly route voice calls, electronic mail and internet-based contacts, and they can connect online purchasers with contact middle agents via chat, escorted browsing or collaborative benign completion. swap providers providing open architecture contact-core products encompass factor Communications Corp., Avaya Inc., Cisco systems Inc. and Nortel Networks Corp.

    The concept is that once contact-center applications are free of single-supplier proprietary programs, consumers can be able to opt for most useful-of-breed products. This open architecture would entice a bigger pool of software developers, leading to new capabilities, lower costs and greater integration.

    as an instance, Nordea AB, a Swedish monetary capabilities business servicing three million shoppers and managing forty four million calls per year (eighty five p.c of them automated), has deployed Genesys SA's Framework to radically change its contact-core operation, enhance the consumer event, and deliver more than US$1.5 million per year.

    Nordea migrated from the ACD purposes on its Nortel switch to Genesys server-based application to manage not simplest cell calls, but furthermore other media. The Genesys IP Contact middle routes contacts and integrates with Nordea's guidance techniques and databases. And rolling the gadget out to further websites changed into rather brief once the initial infrastructure was in location.

    shoppers beget the alternative of which channel they are looking to spend to attain the business, and the Genesys application funnels valued clientele to the foremost attainable and qualified of 1,000 agents in 14 centers, due to the fact language, product/provider and consumer type.

    obstacles, alternatives

    With an open architecture, clients theoretically can combine and robust items from a lot of companies. but that point has not been reached yet. for example, change infrastructure carriers present specialized purposes that are tuned to their personal products or they promote the pre-integrated capabilities of their server purposes - tying collectively ACD, interactive voice response (IVR) and CTI performance. So the world isn't as "open" as it might appear to be.

    On the plus aspect, customers can spend ordinary interfaces to construct integration more convenient. for example, APIs, comparable to TAPI and JTAPI, link switches to servers. CRM and CTI solutions combine with prebuilt adapters, such because the Siemens AG CRM able kit or the Genesys G-Plus adapters. These permit for integration with a true-tier CRM equipment, corresponding to Siebel, in hours or days, in Place of weeks or months. identical connectors combine IVR systems and reporting equipment.

    different outfit additional the goals of openness for ease of integration, interoperability and portability. Standardized yet supple facts schemas and facts access supplied via SQL, Open Database Connectivity (ODBC) and XML are examples. The structure frameworks of Java 2 Platform business edition, .web and customary remonstrate Request broking service structure furthermore are becoming common in touch-middle purposes these days.

    vendors were alert of valued clientele' requisite for convenient integration by means of providing an expanding variety of connectors. Many leverage classic tools, comparable to ODBC; XML; and extraction, transformation and loading (ETL) functions. sooner and simpler integration of the distinctive aspects in a middle is vital to enabling efficient and advantageous aid management.

    Most providers in quest of to integrate with CRM options have, or may have, connectors to the preempt four products (Oracle Corp., PeopleSoft Inc., SAP AG, Siebel systems Inc.) For these carriers, there viable is a full-blown product. For others, it's an open hook - a software structure kit (SDK) or API that can be written to greater easily. CRM and CTI integration now could be tackling the skinny customer structure, which most CRM applications are furthermore migrating to.

    One illustration of a a hit connector method is the AT&T Headend in the Sky (HITS) implementation of Siemens CRM competent package with construct transparent (now Amdocs). The HITS Digital Media core services cable organizations nationwide. The cable businesses desired to route contacts on open concerns to the final agent who handled it if feasible, and pop the quandary ticket on the display. Siemens and AT&T integrated 60 agent pcs in a single weekend, keeping off a mammoth techniques-integration effort.

    Voice self-service apps depart commonplace

    Open architecture furthermore applies to interactive voice response, where one server runs the speech processing for a natural-language consumer interface, and another runs the ordinary VoiceXML code of the core application. both are speaking with typical voice-processing playing cards.

    IVR providers are aiding the VXML average; most are or will quickly be edition 2.0-compliant. This architectural change has big capabilities to open the prior to now small pool of developers and enable portability of purposes, whereas leveraging the net infrastructure and customary facts.

    however, version 2.0 of the VXML universal needs big extensions to assist customer contact functions. CTI-based mostly call control, screen pops and built-in reporting are examples of functions that require capabilities beyond the standard. So a requirements-compliant utility may no longer be transportable throughout systems.

    Many carriers generate VXML code from their proprietary graphical user interface as an choice, easing the migration for these with in-condominium programming potential on their existing platform (however this code is not transportable to a unique platform if it includes any extensions). Some, such as Genesys, are agnostic about requisites. they will drudgery to advertise the construction of the VXML in addition to appellation handle XML specifications. They additionally will conform to and aid Speech utility Language Tags, a competing universal Microsoft is advertising.

    capable, set, do into effect?

    So what's retaining clients again from implementing open utility architectures?

    - Some products are not mature; VXML, as an example, is simply being used through early adopters.

    - Some interfaces don't appear to be basically open.

    - now not enough "canned" interfaces exist, partly as a result of they're dealer-particular, not specifications-primarily based.

    - It will furthermore be complicated to integrate solution elements, potentially resulting in massive device integration costs.

    - The products don't plug and play the way they're marketed.

    - no longer many beget accomplished it before them.

    - upkeep and ongoing administration fees may furthermore be high; as an example, an improve or new liberate in one element can beget a ripple impact to other points for upgrades and checking out.

    As a consumer, you beget to evaluate skills expenses and trade-offs in integration, vicissitude resolution, enhancements etc. Your strategic method to supplier and product alternative needs to correspond with the tradeoffs of integrating most efficient-of-breed products vs. purchasing a collection of products from one vendor.

    Bocklund is vice chairman with forefront Communications. She can be reached at lbocklund@van preserve.internet. -- network World US

    be a piece of the e-newsletter!

    Error: delight determine your electronic mail handle.

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    Citrix deploys Smart Agent technology for Click-to-Call | actual questions and Pass4sure dumps

    Access infrastructure solutions provider Citrix Systems has launched Smart Agent technology with release 6.0 of the Citrix Application Gateway. The Citrix Application Gateway, which provides converged business applications for delivery to screens and speakers of IP telephones, is extending its footprint with Smart Agent to provide productivity applications to users' personal computers in addition to IP telephones. The first Citrix application to spend Smart Agent technology is Click-to-Call which enables telephone users to dial their desk telephone by simply clicking on linked telephone numbers

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    Chapter 1: Cisco Unified Communications Manager Architecture | actual questions and Pass4sure dumps

    This lesson introduces and describes the role, architecture, hardware and software requirements, and the licensing model of the CUCM.

    Chapter Objectives

    Upon completing this chapter, you will beget an understanding of the CUCM architecture and be able to meet the following objectives:

  • Describe the components of a Cisco Unified Communications solution and each component's functionality.

  • Describe the architecture and role of CUCM.

  • Describe the hardware requirements for CUCM.

  • Describe the characteristics of the CUCM operating system.

  • Describe the characteristics of the CUCM database and how it provides redundancy.

  • Describe the licensing model of CUCM.

  • Describe how to calculate, verify, and add license units to CUCM.

  • CUCM Overview

    Cisco Unified Communications (UC) is an IP-based communications system integrating voice, video, data, and mobility products and applications. It enables more effective, secure communications and can transform the way in which they communicate. UC represents a communications paradigm shift dote that of the invention of the telegraph. UC removes the geographic barriers of effective communications through the spend of voice, video, and data integration. business can be conducted with a fluidity that progresses and evolves with you. Information has been at their fingertips for a long time, but UC enables the sharing of this information to create information and value.

    Cisco UC is piece of an integrated solution that includes network infrastructure, security, mobility, network management products, lifecycle services, supple deployment and outsourced management options, end-user and confederate financing packages, and third-party communication applications.

    Cisco UC can drastically change the bottom line of business by creating more effective communications without losing the personal nature of a face-to-face conversation. More effective communication leads to reduced time to market and nimble transformation of business processes through collaboration.

    Cisco UC Solution Components

    The Cisco UC strategy encompasses voice, video, and data traffic within a single network infrastructure. Cisco UC outfit is capable of managing vulgar three traffic types and interfacing with vulgar standards-based network protocols.

    Cisco IP Communications represents a new way of delivering UC functionality to enterprise customers. Instead of delivering a collection of disjointed products with individual release dates, testing methodology, and documentation, Cisco UC is a coordinated release of an integrated set of products that are tested, documented, and supported as a system.

    Figure 1-1 illustrates the four gauge layers of the Cisco UC voice infrastructure model and the components that construct up the layers.

    Figure 1-1

    Cisco Unified Communications Solution Components

    The components of the gauge layers are as follows:

  • Infrastructure layer: The infrastructure consists of routers, switches, and voice gateways. The infrastructure layer carries data, voice, and video between vulgar network devices and applications. This layer furthermore provides elevated availability, management, quality of service (QoS), and network security.

  • Call control layer: The call control layer provides for call processing, device control, and administration of the dial map and features.

  • Call control can be provided by a CUCM, CUCM Express, or CUCM business Edition (CMBE). This bespeak focuses on the CUCM product, which is almost identical to the Cisco Unified CMBE. call processing is physically independent from the infrastructure layer. For example, a CUCM, Cisco Unified CMBE, or CUCM Express in San Jose can process call control for a device physically located in Chicago.

  • Applications layer: Applications are independent from call-control functions and the physical voice-processing infrastructure. Applications, including those listed here, are integrated through IP, which allows the applications to reside anywhere within the network:

  • — Voice mail, integrated messaging, and unified messaging applications are provided through Cisco Unity, Cisco unity Express, or Cisco unity Connections products.

    — Contact centers of various sizes can be built with Cisco Unified Contact focus and Cisco Unified Contact focus Express.

    — Cisco Unified MeetingPlace and MeetingPlace Express are medium- to large-scale conferencing servers that champion video integration. The MeetingPlace product integrates lecture-style conferences with scalable collaboration and control tools. Cisco Unified MeetingPlace Express is positioned to the small to medium-sized enterprises. MeetingPlace Express is the successor of the Cisco Conference Connection server.

    — Cisco Emergency Responder (ER) enhances the existing emergency functionality offered by CUCM. Cisco ER provides physical location updates for mobile devices to guarantee that emergency calls to the public safety answering point (PSAP) are properly routed to the PSAP in freight of emergency calls for that site. Cisco ER identifies the caller location and maps vulgar calls from that physical location to an emergency line identification number (ELIN) through the spend of gauge automatic number identification (ANI)/caller identification (CLID). The ELIN is registered with the PSAP as an Emergency Response Location (ERL). Deploying this capability helps ensure more effective compliance with legal or regulatory obligations, thereby reducing the life and liability risks related to emergency calls.

    — The Cisco Unified Presence server collects information about the availability and communications capabilities of a user and provides this information to watchers of the user as a status indication. The status information includes the user's communications device availability. For example, the user might be available via phone, video, web collaboration, or videoconferencing.

    — gauge protocol interfaces, including Telephony Application Programming Interface (TAPI), Java Telephony Application Programming Interface (JTAPI), Simple remonstrate Access Protocol (SOAP), Q.SIG, H.323, Media Gateway Control Protocol (MGCP), and Session Initiation Protocol (SIP) are available to champion third-party applications.

  • Endpoints layer: The endpoints layer brings applications to the user, whether the conclude device is a Cisco IP Phone, a PC using a software-based phone, or a communications client or video terminal. Cisco UC provides multiprotocol champion for Skinny Client Control Protocol (SCCP), H.323, MGCP, and SIP.

  • Cisco UC Network

    The Cisco UC system delivers fully integrated communications, converging voice, video, and data over a single network infrastructure using standards-based protocols. The Cisco UC system delivers unparalleled performance and capabilities to address current and emerging communications needs in the enterprise environment, as illustrated by the network topology in figure 1-2.

    Figure 1-2

    Cisco UC Network

    The Cisco UC product suite is designed to optimize functionality, reduce configuration and maintenance requirements, and provide interoperability with a variety of other applications. It provides this capability while maintaining elevated availability, QoS, and security.

    The Cisco UC system integrates the following major communications technologies:

  • IP telephony: IP telephony refers to technology that transmits voice communications over a network using IP standards. Cisco UC includes a wide array of hardware and software products such as call-processing agents, IP phones, voice-messaging systems, video devices, conferencing, and many other applications.

  • Customer contact center: Cisco Unified Contact focus products are a combination of strategy and architecture to revolutionize call focus environments. Cisco Unified Contact focus promotes efficient and effective customer communications across big networks by enabling organizations to draw from a broader compass of resources to service customers. These resources involve access to a big pool of agents and multiple channels of communication and customer self-help tools.

  • Video telephony: The Cisco Unified Video odds products enable real-time video communications and collaboration using the same IP network and call-processing agent as Cisco UC. Cisco Unified Video odds does not require special end-user training. Video calling with Cisco Unified Video odds is as smooth as dialing a phone number.

  • Rich-media conferencing: Cisco Unified MeetingPlace creates a virtual meeting environment with an integrated set of IP-based tools for voice, video, and web conferencing.

  • Third-party applications: Cisco works with leading-edge companies to provide the broadest selection of innovative third-party IP communications applications and products focused on captious business needs such as messaging, customer care, and workforce optimization.

  • CUCM Functions

    CUCM extends enterprise telephony features and functions to packet telephony network devices. These packet telephony network devices involve Cisco IP Phones, media-processing devices, VoIP gateways, and multimedia applications. Additional data, voice, and video services, such as converged messaging, multimedia conferencing, collaborative contact centers, and interactive multimedia response systems, interact with the IP telephony solution through the CUCM application programming interface (API).

    CUCM provides these functions:

  • Call processing: call processing refers to the complete process of originating, routing, and terminating calls, including any billing and statistical collection processes.

  • Signaling and device control: CUCM sets up vulgar the signaling connections between call endpoints and directs devices such as phones, gateways, and conference bridges to establish and rend down streaming connections. Signaling is furthermore referred to as call control and call setup/call teardown.

  • Dial map administration: The dial map is a set of configurable lists that CUCM uses to perform call routing. CUCM is responsible for digit analysis of vulgar calls. CUCM enables users to create scalable dial plans.

  • Phone feature administration: CUCM extends services such as hold, transfer, forward, conference, accelerate dial, redial, call park, and many other features to IP phones and gateways.

  • Directory services: CUCM uses its own database to store user information. User authentication is performed locally or against an external directory. Directory synchronization allows for centralized user management. Directory synchronization allows CUCM to leverage users already configured in a corporate-wide directory. Microsoft vigorous Directory (2000 and 2003), Netscape 4.x, iPlanet 5.1, and Sun ONE 5.2 directory integrations are supported. The local CUCM database is a Lightweight Directory Access Protocol (LDAP)-compliant database (LDAPv3) component in the IBM Informix Database Server (IDS).

  • Programming interface to external applications: CUCM provides a programming interface to external applications such as Cisco IP SoftPhone, Cisco IP Communicator, Cisco Unified IP Interactive Voice Response (IP IVR), Cisco Personal Assistant, Cisco Unified Personal Communicator, and CUCM Attendant Console.

  • Backup and restore tools: CUCM provides a disaster Recovery System (DRS) to back up and restore the CUCM configuration database. The DRS system furthermore backs up call details records (CDR), call management records (CMR), and the CDR Analysis and Reporting (CAR) database.

  • Figure 1-3 shows IP phones that logically register with one of the CUCMs in the cluster. Multiple CUCM servers partake one database, and the phone maintains an vigorous connection to both the primary and backup CUCM server. The figure shows the phone's analytic TCP/IP connections to the primary server.

    Figure 1-3

    CUCM Functions

    CUCM Signaling and Media Paths

    CUCM uses SIP or SCCP to communicate with Cisco IP Phones for call setup and teardown and for supplementary service tasks.

    After a call has been set up, media exchange occurs directly between the Cisco IP Phones across the IP network, using the Real-Time Transport Protocol (RTP) to carry the audio. CUCM is not involved in a call after the call has been set up. If the CUCM server were unplugged during the duration of the call, users would not notice unless they attempted to spend a feature on the phone. CUCM is involved only in call setup, teardown, and features. If the CUCM server that set up the call were down during a conversation, conclude users would observe a message indicating "CM Down, Features Disabled" on the LCD screen of the IP phone.

    Example: Basic IP Telephony Call

    Figure 1-4 illustrates a user at phone A placing a call to phone B.

    At the rise of a call, a user at IP phone A picks up the handset, and a message is sent to CUCM letting CUCM know that the device has gone off-hook. CUCM responds to this stimulus by replying with a message that tells the device to play the dial tone file that is stored in the gleam reminiscence of the phone. The user at phone A hears the dial tone and begins dialing the phone number of phone B. SCCP phones forward their digits to CUCM as they are pressed (digit by digit), whereas SIP phones forward their dialed digits in one message (enbloc signaling) by default. SIP phones beget options that allow them to behave similarly to SCCP phones (Keypad Markup Language [KPML] and dial rules). CUCM performs digit analysis against the dialed digits. If a match is found, CUCM routes the call per its configuration. If CUCM does not find a match, a reorder tone is sent to the calling party.

    Figure 1-4

    CUCM Signaling and Media Paths

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