PRESS: TECHtionary.com Announces SIP 2010 Planning Guide

TECHtionary.com Announces SIP 2010 Planning Guide

Two-Five Day Onsite Course called the “Quintessential Guide to SIP”

BOULDER, CO – Techtionary.com (http://www.techtionary.com), the leading provider of SIP & OCS telecommunications training announced today a new leading-edge leading “best practices SIP/VoIP” course called “SIP 2010 Planning Guide” – two-five day onsite training intensive designed exclusively for enterprise IT managers, corporate staff marketing, training business development, sales, channel managers, operations, engineering, support and other corporate managers for VoIP-SIP providers, carriers, manufacturers, and others gain new insights in the SIP-VoIP marketplace. “This course is fills a critical gap in the marketplace for next-gen communications network training,” noted Tom Cross - CEO. “2010 is the major year for the deployment of SIP-Session Initiation Protocol systems globally, this course will provide them with extraordinary tools to help companies succeed,” Cross added.

According to Paul C. Daubitz, President of ATI-Telemanagement (http://www.ati-telemgt.com/), “SIP is the next major significant development for the integration of voice and data. Understanding the critical technical and business issues is crucial for implementation, use and applications development. TECHtionary’s SIP 2010 Planning Guide is the quintessential look at SIP, SIP Trunking, business applications, Security, QoS and more. Scanning the literature, I have found nothing comparable to this research which is presented in such depth in any format.”

This course is produced by TECHtionary.com, the producer of the world’s largest multi-media knowledge library. “By combining the resources and multi-media tutorials and research of TECHtionary this is indepth and critical course on SIP,” noted Thomas Cross – CEO TECHtionary. “After more than three years of research and exhaustive interviews with users, dealers, providers and industry experts, SIP 2010 Planning Guide is the leading major course to address current and future applications.”

Here are just some of the key tutorials in SIP 2010 Planning Guide: - SIP Basics - SIP Trunking - SIP QoS - SIP Firewalls and Security - SIP Applications - SIP TCO-Total Cost of Ownership - Integrated/Converged Access - Key VoIP Options – IAS, Hosted, Managed - Designing and Planning Scalable Global SIP Networks - SIP Total Tutorial with Future Outlook including a look at Microsoft’s OCS-Office Communications Server

This course is also available for private onsite and online delivery.

ABOUT TECHtionary TECHtionary.com produces dealer and customer training programs, online presentations including iPod and PC formats sales brochures, virtual installation manuals and animated online presentations. TECHtionary.com has more than 3,005+ free online animated presentations on data communications, internet, wireless, SIP, OCS-Office Communications Server, VoIP-Voice over Internet Protocol, PBX Systems, central office switching, protocols, telephony, telecommunications, networking, routing, IPTV, WiMax, power systems, broadband, WiFi-wireless fidelity and other related technologies available at http://www.techtionary.com. More details on training are available by calling Tom Cross at 303-594-1694 or cross(at)gocross.com Thomas Cross is a magazine columnist with many key technology publications and CrossTalk was named One of the Top-10 Telecommunications Blogs, http://navigator.cision.com/Top_10_Telecommunications_Blogs.aspx Blog can be found at: http://blog.tmcnet.com/cross-talk/ & Twitter - http://twitter.com/techtionary and a member of the Technical Board of Advisors for the VoIP-Security Alliance.

________________________________________ Effective 1-1-2010 SIP 2010 Planning Guide SIP-Session Initiation Protocol & Optional Introduction to OCS-Office Communications Server 2-5 Days Onsite, Via Webseminar or ~32 hours online. NEW for 2010 - Designing and Planning Scalable SIP Networks (see below for details). © TECHtionary Corporation – all rights reserved ________________________________________ For more information and scheduling, please call Tom Cross 303-594-1694 or cross(at)gocross.com Volume discounts are available. All major credit cards are accepted. Special SIP Forum & Microsoft Partner discounts are available. Additional discounts for multiple sessions are available. For more information on SIP Forum go to www.sipforum.com Reference Promotional Code - SIPF8203 ________________________________________ What Providers, Agents and Users are saying about these courses: According to Matt Jolly IT Consultant, “VoIP Business Executive course VoIP training to a new higher level. For example, there is nothing like the tutorials SIP available anywhere or from anyone. For the channel partner or customer, this course provides critical insights for successful implementation and management. The new user interface speeds learning allowing viewers to grasp complex concepts faster than ever before. With this course, VoIP providers can rapidly accelerate the learning process for their channel partners which in turn accelerate revenues. Now is the time for users and providers alike to make this course an integral part of their business operations.” Matt Jolly - Consultant ............................................ “I am writing regarding Tom Cross and the online training and VoIP/SIP course. By far this is the best program in place, as Tom is one of the most recognized trainers in the United States for VoIP/SIP. It might be worth to add this course to your sign-up package for agents around the country not only as a profit center but something that you could co-brand and have a "leg-up" on your competition.” Bill Bowyer - CEO – VoIP in America .......................................... “The SIP/VoIP courses are more than a superb primer on VoIP/SIP technology; they are an indepth business plan for a VoIP/SIP implementation. In addition, the VoIP/SIP diagnostic and troubleshooting guide is the most thorough approach to VoIP QoS available. I need information that informs but does not overwhelm. Information that guides but not drives you away. The courses provide insights and actionable information that has helped me get the technology we need sooner but saved me a considerable amount in understanding what we didn't need. The SIP course especially is a valuable one which provides much needed information in a readily understandable format.” Paul Daubitz - President – ATI-TeleManagement ________________________________________ Who Should Attend: - This online course is designed for enterprise executive and technical managers, channel partners, VAR-Value-Added Resellers, SI-Systems Integrators, telephone interconnect, agents, master agents and consultants. In addition, this course will benefit corporate technical, staff marketing, training business development, sales, channel managers, operations, engineering, support and other corporate managers for SIP-VoIP providers, carriers, software developers and hardware manufacturers. What You Will Learn: • Review the fundamentals of IP-Internet Protocol and platforms required for high performance SIP-VoIP systems. This includes soft switches, gateways, routers, services and other critical components. • Explore business applications and opportunities. Review what customers are buying today and why they are buying. In addition, emerging “killer applications” will be explained in depth. • Quickly grasp complex subjects such as H.323, MGCP and SIP. As SIP-Session Initiation Protocol emerges are the key VoIP communications protocol, discover how this technology will impact all voice communications systems from key, PBX, IP-PBX, hosted, managed and other systems. • Understand basic and advanced SIP-VoIP concepts features. From hosted, managed, IAS, and IP-PBX, quickly understands “what’s-what” for different customer applications and business models. • Probe the issues behind Integrated and Converged Access. Understand when and why organizations need a converged access solution. • Understand why “network assessment” is critical to any SIP-VoIP implementation and why this step cannot be overlooked. • Address the issue of QoS-Quality of Service by overcoming jitter, echo, noise and other network problems. Review the role of RTCP and other tools to monitor and maintain high performance VoIP networks. • Understand the functions of the new communications “toolbar.” See how the benefits of “unified communications” as they improve business operations. • Assess the Top-10 issues why SIP trunking and hosted VoIP is more than “dial-tone,” and how it can represent change in the business and business model of even the smallest enterprises. Discuss and explore new ways to improve fundamental business processes. • Explore how a SIP-VoIP call is processed and review potential security attacks. Discover how SPIT, VOMIT, DOS and other terrorist attacks can target not just data, but voice packets. • Review SIP and SIP Trunking and all the implications and applications from TCO-Total Cost of Ownership to QoS-Quality of Service. SIP Trunking is the most profound new form of telecommunications since POTS. • ? Explore Microsoft’s OCS-Office Communications Server features, concepts, call flows, configurations and other issues for evaluation and implementation. ________________________________________ Detailed Course Outline Day One - Fundamentals of Data/Internet Telecommunications NOTE: Day one is recommended for those wanting an update in internet technologies, data networking and network services (bandwidth). - Fundamental Network and IP Technologies – the IP in SIP • 1 – Voice-to-Digital-to-Packet Transmission • 2 – Back To Basics – Cabling, Conduit and Electrical Systems • 3 – Transmission Concepts – DSL, T-1/E-1, ISDN-PRI, SIP Trunking, GIG-E • 4 – Optical Fiber & Bandwidth • 5 – Integrated Access Services – Dynamic Bandwidth Allocation – BOD-Bandwidth On Demand • 6 – Introduction to IP-Internet Protocol and VoIP-SIP , MPLS-Multi-Protocol Label Switching, DiffServ-Differentiated Services, DSCP Differentiated Services Code Points and Packet Priority Classifications, TOS-Type of Service, EF-Express Forwarding, MPLS Uniform mode, MPLS Pipe and Short-Pipe modes, WRR-Weighted Round Robin, TCB-Traffic Conditioning Blocks - Marker, Meter, Shaping, Droppers and PHB-Per Hob Behavior. • 7 – TCP/IP and other Protocols and Layers – RTP, RTCP, SDP, SOAP, SALT - Call processing with Route, Image, DHCP, DNS, Image, Configuration servers • 8 – Hardware – Routers, Switches – MAC-Media Access Control, WiFi-VLANS-VPNS • 9 – Protocols “Rules of the Road” – H.228, H.323, MGCP, SIP, and Desktop “Softphones,” “Toolbars” and other end points (desksets) • 10 – IP-PBX and Hosted VoIP/SIP – Integrated/Unified/Homogenized - Top-10 Critical Technologies to SIP 1 – IP protocol, IP networking and a VPN 2 - The difference between IAS-Integrated Access Service versus Converged Access Service - Enhanced IAS with MPPP-Multi-link Point-to-Point Protocol, PPP Multilink Protocol (MP), L2TP-Layer 2 Tunneling Protocol - VPLS-Virtual Private LAN Service - new name for metro-ethernet - Switching Versus Routing - key benefits of VPLS 3 – SIP-Session Initiation Protocol Trunking - - SIGTRAN (Signaling Transport) - SCTP-Stream Control Transmission Protocol 4 – Justification for an IP PBX – options and approaches 5 – Technical difference between IAS-Integrated Access Service, Hosted and Managed VoIP - Call processing with Route, Image, DHCP, DNS, Configuration servers - Media Gateways replace PBXs - the following tutorials are some examples of customer applications of MG-Media Gateways: - Connection of IP-PBX to PSTN - Connection of IP-PBX to PSTN & SIP trunk provider - Survivable connection to SIP trunk provider - Connection of PBX & IP-PBX to PSTN & SIP trunk provider - Connection of IP-PBX to Hosted VoIP provider - Connection of IP-PBX & PSTN to Microsoft OCS Server - SC-Session Controllers or SBC-Session Border Controllers are access devices operate at Layer 5 Session Layer, where as routers operate at Layer 3 Network. Some of the key SBC/SC functions are: - Secure network peering - private and public to enhance performance - Topology hiding - using various types of inter-AS-Autonomous System features as well as separating media (voice) and hide signaling (IP addresses) and data streams (traffic) - Border call routing - routing at AS level rather than with interior protocols - Interoperability - access/restrict to reduce voice spam - QoS & Call Admission Control - load/jitter correction - Billing systems interoperability - reduce billing errors - NAT-Network Address Translation - routing for maximum performance - CALEA-Communications Assistance for Law Enforcement Act - Compatibility with billing - Dialect conversion - Protocol conversion - Codec conversion - Firewall restrictions - Wholesale and Transit peering 6 – “Open Source” PBX options 7 – QoS-Quality of Service importance - how to measure it and fix it 8 – Softphones – Where they make sense - user benefits 9 - The difference between IPT-Internet Protocol Telephony and VoIP - Cisco, Broadsoft, Sylantro and other platforms 10 - Unified Communications – Mobility Applications ________________________________________ Day Two - Introduction to SIP-Session Initiation Protocol - SIP Planning - SIP Introduction and Overview - SIP Definition – IETF (RFC-3261) and Manufacturers - CPL-Call Processing Language - AOR-Address Of Record – q-values - Location Service - DNS-Domain Name Service - CPL-Call Processing Language - B2BUA-Back-2-Back User Agent - Session Initiation Protocol for Telephones (SIP-T): RFC 3372 - SIP-SS7-Signaling System 7 call processing including – IAM-Initial Address Message, Routing label, CIC-Circuit Identification Code and Message Type Code. Examples of Message Type Codes include: Called Number, Calling Number, DPC-Destination Point Code, OPC-Origination Point Code, SS7-ISUP ACM-Address Complete Message, ANM-ANswer Message, CPG-Call ProGress Message, COT-COTinuity Message, SUS-SUSpend Message, RES-RESume Message, FOT-FOrward message Transfer, INR-INformation Request message, INF-INFormation Message, RELease and other messages. - SIP – Applications Layer 7 Protocol – Peer-to-Peer protocol - SIP – Before and After - SIP and Hosted – Better or Worse or Both - SIP Signaling – Introduction, URI-Uniform Resource - SIP & SBC-Session Border Controllers, servers, gateways, - SIP with and without IADs-Integrated Access Devices - SIP and SIP Phones, Softphones, Mobility, - SIP Signaling Basics – Inbound/Outbound calling - UC-SIP Bandwidth Planning - Critical Concepts for PC Video, data and voice - SIP Trunking - Four types and counting of SIP Trunking offerings - SIP Trunking – Incremental “Slope” Growth - CODECS-COmpression-DECompression signal processors – issues and answers - SIP Trunk Replacement & Disaster Planning - SIP & Open Standards - SIP and Trunk Replacement – same or different thing - SIP and Proxy ARP-Address Resolution Protocol - SIP and HSRP-Hot Standby Routing Protocol - SIP and MPLS-Multi-Protocol Label Switching – COS and QoS - SIP QoS – oxymoron or critical concept - SIP on-net and off-net issues – overflow call processing - SIP TCO-Total Cost of Ownership – Top-10 Benefits - SIP Technology - Indepth - SIP – OSI-Open Systems Interconnection - "If you do not know where you are going, what difference does it make which path you take".....Cheshire Cat (Alice in Wonderland) - SIP “Methods” – Writing call processing as easy as email – invite, ACK, bye, etc. - SIP Signaling “commands” – 1xx-6xx - SIP Inbound and Outbound call processing - SDP-Session Description Protocol - headers, Via, Max-Forwards, To:, URL-Uniform Resource Locator, URI-Uniform Resource Identifier, call-ID, Cseq, Contact, Content-Type, Content-Length, Security and others - Session Description Protocol Security Descriptions (SDES) - SIP Features - Forks, SIP Proxy, Redirect, Presence, Forking – parallel-sequential-mixed, loops, spirals - SIP Network devices - UA-User Agent, UAC-User Agent Client, UAS-User Agent Server -Proxy Server, Redirect Server, Registrar Server, B2BUA-Back-to-Back User Agent - SRTP-Secure Real-time Transport Protocol (RFC-3711) - Authentication Tag and the Master Key Identifier - Encryption

________________________________________ Day Three - Advanced SIP Planning and Security - SIP Security – “Best Practices” – Reality Check - SIP Security “Best Practices” – overview - SIP Firewalls and Security – SPIT-SPam over Internet Telephony, DOS-Denial Of Service, VOMIT-Voice Over Misconfigured Internet Telephony and other emerging problems - SIP Security and “Access Policy” – Stateful IP Filtering and Inspection, Static and Stateless IP Filters, TLS-Transport Layer Security, NAT-Network Address Translation, Persistent connection, Multi-homed hosts, etc. - SIP and MIM-Man-In-the-Middle attacks – Understanding wireline and WiFi wireless attacks - Telephone Numbers – North American Numbering Plan and International ENUM-E.164 - SIP Security Architectures – Building Blocks - SIP Security Architectures – eight different VoIP configurations evaluating SIP-Aware Firewalls and other security options: - Type 1 – Dedicated IP Pipe for VoIP - Type 2 – Merged MPLS-Pipe with LER Tagging VoIP - Type 3 – Merged IP pipe with SIP-Aware Firewall (SAFW) - Type 4 – Separate IP Pipe for VoIP with Existing Non-SIP Firewall and SIP-Aware Firewall (SOFW) - Type 5 – Merged IP Pipe with Incumbent Non-SIP-aware Firewall, No DMZ Port and SIP-aware Firewall (SAFW) - Type 6 – Looks like Type 5 but Merged IP Pipe with Incumbent Non-SIP-aware Firewall, No DMZ Port and SIP-aware Firewall - Type 7 – Merged IP Pipe with Incumbent Non-SIP-aware Firewall with a DMZ Port - Type 8 – Merged IP Pipe with Incumbent Non-SIP-aware Firewall - Other approaches to SIP Security - Proxy/Gateway Inside the Firewall, Proxy/Gateway in Co-Edge Mode and Proxy/Gateway Outside the Firewall, how Firewalls add time delays to TCP/IP - 50-Point Comprehensive SIP Security Checklist - more than 50 different security concepts to review and include in the implementation and ongoing network management program - SIP Security-Privacy Lifecycle Management - key planning for capturing, storage, users, and disposition (archiving/destruction) NEW for 2010 - Designing and Planning Scalable SIP Networks The five areas of discussion presented (with modifications I added) are: Here are the highlights in text format: 1 - Benefits of Server Virtualization aka Abstraction - Distributed user endpoints and application servers - Scalable growth with QoS control abstraction - MACS* via signaling "abstraction" *Moves, Adds, Changes & SAS-Stand Alone Survivability - SAS enables backup for SIP devices by the multiple "abstract" local or cloud Media Gateway(s) 2 - Benefits of SIP Trunking - Session and Signaling control layers for: - On-net (IP PBX to IP PBX) - Off-net (IP PBX to local PSTN) - Off-net (IP PBX to LD/IDDD PSTN) - SPOC/SMOC-Single/Multiple Point of Connection - National control points for access (vendors) - Migration path to ALL-SIP & IMS-3GPP 3 - Benefits of Number Transparency - End-to-end media control (transparency) for voice or other media types - Access to SS7 applications for text messaging, mobility or call center routing - Enterprise control over call completion whether PSTN or SIP rather than provider - Migration path to E.164 & IMS-3GPP 4 - Benefits of SIP Routing - Eliminates "mesh" network mess - Add intelligent Session Layer 5 routing - Maintains end-to-end media transparency - Add media CoS-Classes of Service - Adds SPOC NOC-Network Operations Center - Migration path to ALL-SIP & IMS-3GPP 5 - Benefits of SIP Proxy Peering Networks - Platform for multi-vendor services - Ensures multiple CoS & QoS options - Migration path to ALL-SIP & IMS-3GPP - Foundation for any future network needs - SIP Class of Service & Quality of Service - SIP COS-Class Of Service and QoS-Quality of Service – ethernet meets “smart” IP - Managing “real-time” voice with RTCP-Real-Time Control Protocol – MRB-Metrics Report Blocks - Inside MRB – what’s what with all the info - SIP Applications and Future Outlook - SIP Applications – IM-Instant Messaging call screening - SIP Presence Communicated by IM-Instant Messaging - Click-to-call and others - SIP for Call Centers – calling options and pricing benefits - Event Notification - Ondemand Conferencing - Integration of additional "third-party" developed SIP-enhanced services provides additional business and enterprise justification for SIP trunking. - UDDI-Universal Description, Discovery and Integration uses standards-based services such as XML, HTTP, SOAP, TCP/IP (define above) uniform service description and service discovery protocol. Discovery services provide a consistent publishing interface and allow programmatic discovery (registration) of services. Description services provide the means for internet registration - to be found but not advertisement or placement on search engine listings. UDDI file structures are designed using a "publish-once-read-by-many" concept. That is, web site URL-Uniform Resource Locator can be designed using UDDI standard file structures which can be published to the UDDI server network. The UDDI network can be accessed (discovered) by search engines, customers and other list builders in a standard published (register) format. UDDI Registries and protocol servers with: - White Pages - Names, Address, Contact and Vcard information - Yellow Pages - Industry categorizations and taxonomies - Green Pages - Technical information including internal URL file discovery structures - UDDI is also designed to replace the robot.txt search engine web site document structure concept. Here are some of the web site description-discovery-registry information retrieved by search engine spiders/bots and other retrieval programs. - Voice-driven yellow pages - SALT-Speech Applications Language Tags adds voice commands to web applications. SALT is an extended set of markup (meta) tags based on XML-eXtensible Markup Language though compatible with HTML-Hyper-Text Markup Language and others. - SIP – exciting new applications - SIP Total Tutorial with Future Outlook – IMS-IP Multi-media Systems – content servers, wireless integration, media gateways, etc. - Top-10 Steps to a Successful SIP Implementation 1 - User Needs Assessment 2 - Network Assessment 3 - Systems Upgrade - Indepth POE-Power Over Ethernet & Comprehensive Disaster Planning Tutorial 4 - Pre–Installation Planning 5 - Data Systems Integration - VLANs, VoWLANS, Planning for WiFi, WiFi and IP Wireless "Roaming," WiFi Security and more 6 - Installation and Cutover 7 - Managing Change - Training 8 - Ongoing Use and Expectations 9 - Billing 10 - Managed Services - TCO-total cost of ownership, monitoring, remote support, training, business development and others & Future Applications - Diagnosing & Tools for Troubleshooting SIP Networks 1 - Problems: - Delay - Jitter - Equipment Configurations - Clipping & Dipping - VAD-Voice Activity Detectors - - Connection Issues - Echo - Signal-Noise Level and & Loss - Comfort Noise - Packet Loss Concealment - Zero Insertion - Waveform substitution - Model-based methods - Crosstalk - Nearend and Farend - Serialization - Packet Payload Delays - Packet Sizing Problems - Take the "Vo-eye-P Test" - Transcoding Problems - Asynchronous Transcoding Problems - Electrical Interference - Surges, Sags, Shared Neutrals 2 - Testing for Problems - RTCP-XR-MRB-eXtended Reports - Metrics Report Block 3 - More than 30 Problems & Solutions - like "CarTalk" bring your problems to "Nettalk" 4 – Best Practices - review of concepts such as Resiliency & Reliability – QoS in VoIP-SIP 5 – Vendors of Technical Solutions for VoIP Network Management 6 - Conclusions and the Bottom Line NOTE: Course contents are constantly being update, please inquire about special requirements. ________________________________________ Day Four - (with optional add-on day of labs) - Introduction and Demonstration of OCS-Office Communications Server Here are just some of the terms used in Microsoft OCS-Office Communications Server. Each of these terms and others are explained in this OCS Special Edition: • Architecture – Front End, AD-Active Directory, Conferencing, Perimeter and Applications Servers • PBX Co-existence • UC-Unified Communications • Advanced Telephony Features • PBX Integration Initiative – UC-Compatible, Non-UC-compatible • Mediation Server – Forking • SIP – Microsoft SIP “Standard” – Inbound/Outbound Calls, OnHold, Making a Call Via a PBX, Answering a Call Via a PBX, Call Forwarding – Always, DND-Do Not Disturb, Call Transfer, 3-Way Conferencing • SIP-to-PBX • SIP-to-PSTN • Microsoft Management Console (MMC) • Administrative Tools "Snap-in" • Call Detail Records • Phone Management and Usage • IM-Instant Messaging, Chat, Chatrooms, etc. • Video Conferencing "Roundtable" - Planning and Compliance • Saving of IM conversations for compliance • Planning for OCS • Troubleshooting • Trunking – bandwidth, delay, echo • Telephone deskset issues • IP routing check • PBX interoperability • Third-party integration – open discussion • LCS-Live Communications Server and OCS coexistence NOTE: This session is constantly changing due to new releases coming from Microsoft such as Wave 14 expected for general release in 2/2010. In addition, there may will be "LIVE" demonstration of IM, video conferencing, call processing and other features from a live OCS system. NOTE: This topics presented may not be the complete list of issues discussed at time permits. An optional five-day or half-day is available for custom training. Microsoft is a registered trademark of Microsoft Corporation. ________________________________________ Course Leader - Thomas B. Cross – CEO TECHtionary.com has three decades of experience in startups and consulting advisor with leading providers and venture capital companies in market planning and development, hardware/software design and development, project management, intellectual property in telecommunications, information technology, conferencing, teletraining, telecommuting, groupware, networks, call centers, internet, artificial intelligence and other fields. He has managed the successful development of more than 10 software, hardware and internet products to market and received industry awards for this work. He has authored 13 books, wrote, produced and directed 15 commercial videos and creator and producer of the World's Largest Animated Knowledge Source on Technology – http://www.techtionary.com – recipient of Web Hosting Magazine Editors Choice for Best Technical Help. Tom is a columnist for many leading publications such as Internet Telephony where he is the Technology Editor and columnist on OCS-Office Communications Service Newsletter with a blog at http://blog.tmcnet.com/cross-talk/. He is a member of the Technical Board of Advisors for the VoIPSA-VoIP Security Alliance. Tom holds CompTIA Certified Security Professional certification and Pearson Vue Certified Test Administrator. Tom is also CEO and Managing Editor of OCS Forum – http://www.ocsforum.com

Scheduling & Classroom requirements To confirm delivery dates, fees including estimated travel and expenses are required. Purchase Orders (P.O.) are accepted but do not confirm dates. Final expenses are due within 15 business days after completion of class. Please note that payments received for delivery dates are subject to final approval by Techtionary. Final student attendance is set 30 days prior to the first day of class with additional student payment due on the first day class. Techtionary will provide access to the online course at no extra charge. Client will provide room, computers for lab access, high-quality video projector, desks, power strips, beverages, food service and other amenities. Classes are conducted each day (subject to mutually agreeable changes) from 0900-1600 with one hour for lunch. Access to training room will be available from 0800-1700 each day.

Course content and terms are subject to change without notice. All course content delivery may not be completed due to time restraints such as student questions and special explanations of concepts presented. Content is wholly-owned by Techtionary Corporation, a Colorado corporation which provides invoice/billing. Corporate and shipping address is 2525 Arapahoe E-4-313, Boulder, Colorado 80302.

TECHtionary.com produces dealer and customer training programs, online presentations including iPod and PC formats sales brochures, virtual installation manuals and animated online presentations. TECHtionary.com has more than 3,005+ free online animated presentations on data communications, internet, wireless, SIP, OCS-Office Communications Server, VoIP-Voice over Internet Protocol, PBX Systems, central office switching, protocols, telephony, telecommunications, networking, routing, IPTV, WiMax, power systems, broadband, WiFi-wireless fidelity and other related technologies available at http://www.techtionary.com. More details on training are available by calling Tom Cross at 303-594-1694 or cross@gocross.com. Thomas Cross is a magazine columnist with many key technology publications and a member of the Technical Board of Advisors for the VoIP-Security Alliance. He holds Comptia Security+ and other certifications.

OCS Forum provides classroom and webseminar training as well as a non-production environment for those IT departments without additional equipment, budget or time. This allows planners and users to test ideas, dial in and dial back out, IM file transfers, remote desktop sharing, video conferencing, run scenarios, review logs, break linkages and learning about new telephony features and network access. OCS Forum is also designed for both the system integrator/consultant who wants to learn about OCSR2 without having to build their own system as well as the enterprise customer who doesn't have the time, resources or knowledge to develop one. OCS Forum Labs are designed to be "hands-on" or "over-the-shoulder" with experts available for Q&A and classes for feature-specific review. About OCS Forum OCS Forum is a vendor-independent laboratory environment designed for learning, technical guides, knowledge resources and online "live" services. OCS Forum provides planning, project management, consulting, training, case studies, white papers, speaking engagements, market/customer research, network planning and other services. For more go to http://www.ocsforum.com

OCS Forum Expo 2010 will be held in Boulder, Colorado, June 15 & 16 at the St. Julien Hotel (www.stjulien.com) with keynotes from OCS MVPs, Media Gateway/SBC experts, customers, planners and others. Attendees and exhibitors will receive thousands of dollars in valuable videos, training courses and online access to a "live" OCS system. There will be sessions on OCS Planning, QoS, Security, Firewalls, Media Gateways, PBX Integration, Mobile OCS, Communicator, WAN-Bandwidth Planning, Customer Applications, Troubleshooting, Session Border Controllers and others. For prospectus, speaker, exhibitor, information, please contact Tom Cross cross@gocross.com or 303-594-1694.