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Chief expertise Officer and govt Editororiginally posted on VoIP & gadgets blog, right here: http://blog.tmcnet.com/blog/tom-keating/asterisk/asterisk-with-office-communications-server-2007.asp.
Asterisk (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 answer (Asterisk) with Microsoft's proprietary OCS 2007 UC platform could be well-known, but it surely is. basically, I even have 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 idea 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 do 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 aware about a number of Fortune 1000 corporations, including security-mindful predominant banks, that have deployed OCS 2007.
happily, no longer all within the Asterisk community have 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, name handle, IM, collaboration, video, and extra. right here's a pattern of me using it.
Of path, which you could use X-Lite, a free SIP client, with Asterisk, nonetheless it's functionality and integration is rarely as feature-rich as the whole OCS platform. though if cost is your only issue, Microsoft's OCS licensing may be too a whole lot for some.
So in any adventure, what is the issue in integrating Asterisk with OCS 2007? well, as I even have 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 prevent this notwithstanding, considering any future updates might negate the patch and probably ruin the "legitimate" Asterisk. further, SIP over TCP isn't very normal (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 use OpenSER as a SIP UDP to SIP TCP gateway as neatly. really it seems like 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 use the good value pbxnsip equipment as an inexpensive SIP UDP to SIP TCP gateway in case you need and use this along side Asterisk. definitely, pbxnsip will also be with ease integrated OCS 2007 and with change 2007 UM (Unified Messaging). try my full 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 offer Asterisk to OCS integration, nonetheless it seems like a work in development.here's strictly from the rumor-mill, but I heard that SIP over UDP support "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 have efficiently gotten it to work at once with OCS Mediation server. So in spite of the fact that Microsoft would not offer SIP over UDP assist in the next release, it feels like Asterisk is adding SIP over TCP support.
I should still point out that Asterisk lacks CSTA support and alas OpenSER does not help 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 answer I discovered that adds rudimentary CSTA capabilities to Asterisk is offered with the aid of M-Networks. Their solution, known as the Unified name handle Gateway (information - Alert), services as an intermediary that enables the Microsoft are living 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 use their desktop as a cell. while it become designed for LCS 2005, in thought the application could work on its successor, OCS 2007. here's a few of its facets:
it be critical to factor out that the CSTA common provides for third party CTI applications together with superior ACDs and IVRs which makes use of CSTA for speaking with a PBX. this is vital in enormous name 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 name handle and might be concept of as a CSTA/TSAPI equivalent. Some Asterisk-based mostly options offer TAPI-primarily based computing device name control as smartly, but TAPI may not help you with OCS 2007 integration - exceptionally with the OCS Communicator presence/VoIP client (shown to the right).
Let's seem 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 send the repute of the SIP purchasers to OCS crucial for presence and other performance. It cannot for example be aware of when a specific session has been terminated. Asterisk in spite of this is dialog stateful and is privy to all classes constantly. With uaCSTA interaction between OCS 2007 and Asterisk the consumer-state and the VoIP-cell-state will also be exchanged.The large query is will anybody in the Asterisk group write the code to do CSTA-to-AMI? anyone?
Of direction, if you have to have CSTA for the superior telephony features today, you might seem right 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 all with their IP-PBX - identical goes for Nortel. I believe SIP/CSTA options which are "IP-PBX agnostic" are also obtainable from Audiocodes, Dialogic, and Quintum.
I should still aspect out that OCS truly makes use 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 help purposes. uaCSTA can be applied through a couple of different types of SIP user brokers, together with directly by way of a SIP user agent on a SIP cellphone. uaCSTA can also be carried out by using a SIP B2BUA to augment third-celebration name 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 work 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 LiteScape.com, 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 get it to work. extra, in case you need superior points like RCC and twin-forking, you'll need 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 have efficaciously gotten it to work 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 have uaCSTA guide very shortly. besides the fact that children pbxnsip is rarely Asterisk, it's yet another open source IP-PBX answer 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 run on multiple operating systems, including Windows, Linux, etc.]
here's a screenshot of the leading step performed by means of the pbxnsip web admin:
[ Back To TMCnet.com's Homepage ]
graphic this: instead of having to operate an expensive name core tied to a actual location, you may have created a digital, multimedia contact center staffed by using brokers working from home or in distant workplaces, linked through a voice-over-IP community. you might have launched cash-saving, self-carrier consumer purposes based on standards corresponding to XML and VoiceXML that cut back the weight to your agents. you could have 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 form of monitor pops.
consumers attain your contact core any approach they need - voice name, electronic mail,chat and precise-time collaboration over the web. The client event is constant across all media, and each interaction is immediately mentioned so updated information is obtainable right away. The contact middle blends supplies, and manages and studies on all contacts via a standard 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 critical 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 need to migrate call routing and reporting, traditionally handled through automatic call distributors (ACD), from closed switching systems to open, industry-typical servers.
Open structure offers the promise of infrastructure-agnostic purposes. Routing, reporting and management capabilities can run 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 kind 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 save 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 also 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 have the alternative of which channel they are looking to use 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.
With an open architecture, clients theoretically can combine and healthy 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 seem to be.
On the plus aspect, customers can use ordinary interfaces to make 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 equipment additional the goals of openness for ease of integration, interoperability and portability. Standardized yet flexible 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 Object Request broking service structure also are becoming common in touch-middle purposes these days.
vendors were aware of valued clientele' need for convenient integration by means of providing an expanding variety of connectors. Many leverage normal 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 appropriate four products (Oracle Corp., PeopleSoft Inc., SAP AG, Siebel systems Inc.) For these carriers, there possible is a full-blown product. For others, it's an open hook - a software building 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 also 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 make clear (now Amdocs). The HITS Digital Media core services cable organizations nationwide. The cable businesses desired to route contacts on open concerns to the last 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 big techniques-integration effort.
Voice self-service apps go commonplace
Open architecture also 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 large 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 general needs large 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 work to advertise the construction of the VXML in addition to name handle XML specifications. They additionally will conform to and aid Speech utility Language Tags, a competing general Microsoft is advertising.
capable, set, put 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 seem 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 also 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 have accomplished it before them.
- upkeep and ongoing administration fees may also be high; as an example, an improve or new liberate in one element can have a ripple impact to other points for upgrades and checking out.
As a consumer, you have to evaluate skills expenses and trade-offs in integration, difficulty resolution, enhancements etc. Your strategic method to supplier and product alternative needs to agree 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 USbe a part of the e-newsletter!
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Download the authoritative guide: Cloud Computing 2019: Using the Cloud for Competitive AdvantageThe new Sun Certified Solaris Security Administrator exam is now available in beta format. The beta period will last until March 8.
Exam objectives will be sent to those you register for the beta at Sun's Web site.
Microsoft Offers Free .NET Exam to Current MCSDs
Starting March 1, MCSDs who certified on the Visual Studio 6.0 track can get a free voucher good for exam 70-300 from the MCSD for .NET track. Exam 70-300: Analyzing Requirements and Defining .NET Solution Architectures is one of four core exams required for MCSD for .NET certification. It usually costs $125. To date, more than 41,000 people have earned the MCSD title. Exam 70-300 went live Feb. 10. The free voucher must be used by Sept. 30. Existing MCSDs can apply for a voucher through the MCP Secure Web site. For more details on MCSD .NET visit GoCertify's MCSD .NET page.https://o1.qnsr.com/log/p.gif?;n=203;c=204660774;s=9478;x=7936;f=201812281339040;u=j;z=TIMESTAMP;a=20403972;e=i Marimba Certification Program Reborn
Marimba's certification program has emerged from its redesign with two certifications that IT professionals can earn. The first, Certified Marimba Professional (CMP), requires passage of 2-4 online exams, depending on your choice of track. Options include Desktop Management Administrator, Server Management Administrator, Application Packager, and System Architect.
The second level of Marimba certification, Certified Marimba Consultant, requires that you first earn CMP certification, plus pass a hands-on practical exam that lasts one day. For more information see GoCertify's Marimba page.
SAS Base and Advanced Programming Exams Live March 29
The final versions of the SAS Base Programming and SAS Advanced Programming exams are scheduled to go live in their final (non-beta) format by March 29. Unlike earlier SAS exams, these are not version-specific. This reflects the new organization of the SAS certification program, in which certifications are designed around job role levels. For more information on all available SAS certifications see our SAS page.
Nortel Networks Releases Two New Design Specialist exams
You can now earn Nortel Networks Design Specialist (NNCDS) certifications in VoIP CallPilot 2.0 and SCCS/Express/TAPI/Web Client/Agent.
The new CallPilot 2.0 (#920-155) exam is one of three required to earn NNCDS - VoIP CallPilot 2.0 certification. Completion of the VoIP Technology (#920-801) and Meridian 1 Database (#920-318) exams is also required to earn VoIP certification.You can earn NNCDS - SCCS/Express/TAPI/Web Client/Agent certification by successfully completing the new SCCS/Express/TAPI/Web Client/Agent (#920-316) exam. The exams are proctored by Prometric.
That's all for this edition of Certification Watch. Please keep your certification news and tips coming to email@example.com.
Anne Martinez is the author of Cheap Web Tricks: Build and Promote a Successful Web Site Without Spending A Dime and Get Certified and Get Ahead. She also is the founder of GoCertify.com.RELATED NEWS AND ANALYSIS
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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 use 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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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 have 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.
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 we communicate. UC represents a communications paradigm shift like that of the invention of the telegraph. UC removes the geographic barriers of effective communications through the use of voice, video, and data integration. Business can be conducted with a fluidity that progresses and evolves with you. Information has been at our fingertips for a long time, but UC enables the sharing of this information to create knowledge and value.
Cisco UC is part of an integrated solution that includes network infrastructure, security, mobility, network management products, lifecycle services, flexible deployment and outsourced management options, end-user and partner 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 equipment is capable of managing all three traffic types and interfacing with all 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 standard layers of the Cisco UC voice infrastructure model and the components that make up the layers.
Cisco Unified Communications Solution Components
The components of the standard layers are as follows:
Infrastructure layer: The infrastructure consists of routers, switches, and voice gateways. The infrastructure layer carries data, voice, and video between all network devices and applications. This layer also provides high 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 plan and features.
Call control can be provided by a CUCM, CUCM Express, or CUCM Business Edition (CMBE). This book 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 Center and Cisco Unified Contact Center Express.
— Cisco Unified MeetingPlace and MeetingPlace Express are medium- to large-scale conferencing servers that support 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 charge of emergency calls for that site. Cisco ER identifies the caller location and maps all calls from that physical location to an emergency line identification number (ELIN) through the use of standard 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.
— Standard protocol interfaces, including Telephony Application Programming Interface (TAPI), Java Telephony Application Programming Interface (JTAPI), Simple Object Access Protocol (SOAP), Q.SIG, H.323, Media Gateway Control Protocol (MGCP), and Session Initiation Protocol (SIP) are available to support third-party applications.
Endpoints layer: The endpoints layer brings applications to the user, whether the end device is a Cisco IP Phone, a PC using a software-based phone, or a communications client or video terminal. Cisco UC provides multiprotocol support for Skinny Client Control Protocol (SCCP), H.323, MGCP, and SIP.
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.
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 high 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 Center products are a combination of strategy and architecture to revolutionize call center environments. Cisco Unified Contact Center promotes efficient and effective customer communications across large networks by enabling organizations to draw from a broader range of resources to service customers. These resources include access to a large pool of agents and multiple channels of communication and customer self-help tools.
Video telephony: The Cisco Unified Video Advantage products enable real-time video communications and collaboration using the same IP network and call-processing agent as Cisco UC. Cisco Unified Video Advantage does not require special end-user training. Video calling with Cisco Unified Video Advantage is as easy 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 critical business needs such as messaging, customer care, and workforce optimization.
CUCM extends enterprise telephony features and functions to packet telephony network devices. These packet telephony network devices include 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 all the signaling connections between call endpoints and directs devices such as phones, gateways, and conference bridges to establish and tear down streaming connections. Signaling is also referred to as call control and call setup/call teardown.
Dial plan administration: The dial plan is a set of configurable lists that CUCM uses to perform call routing. CUCM is responsible for digit analysis of all calls. CUCM enables users to create scalable dial plans.
Phone feature administration: CUCM extends services such as hold, transfer, forward, conference, speed 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 Active 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 also 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 share one database, and the phone maintains an active connection to both the primary and backup CUCM server. The figure shows the phone's logical TCP/IP connections to the primary server.
CUCM FunctionsCUCM 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 use 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, end users would see 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 beginning 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 flash memory of the phone. The user at phone A hears the dial tone and begins dialing the phone number of phone B. SCCP phones send their digits to CUCM as they are pressed (digit by digit), whereas SIP phones send their dialed digits in one message (enbloc signaling) by default. SIP phones have 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.
CUCM Signaling and Media Paths
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