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Sipxecs Enterprise Communications System sipXecs Enterprise Communications System A complete IP PBX for today and tomorrow Key Attributes that is easy to use, fully featured and robust sipXecs offers: Voice Mail Integrated voice mail system with personal sipXecs Enterprise Communications System (ECS) is a highly robust and auto-attendant per user scalable, enterprise-grade, and fully featured unified communications solution with integrated voicemail, unified messaging, presence, call Automated Call Distribution Integrated Call Center solution that center, multiple auto attendants, paging and intercom services and a distributes calls to multiple agents and powerful plug & play web-based configuration and management system. queues through intelligent routing sipXecs is based entirely on the Session Initiation Protocol (SIP) standard and operates on standard Linux operating system. It interoperates with a Unified Messaging Voice mail messages can be retrieved by large number of third party phones, PSTN/IP gateways and applications web browser or forwarded to any email such as Microsoft Exchange 2007 without compromising ease of use. client. sipXecs is a 2nd generation voice over IP (VoIP) open source system not Multiple Auto Attendants Auto attendants are easily configured via built for geeks but enterprise users requiring easy to use, stable and browser interface. scalable solutions for mission critical applications. Thousands of small and large enterprises rely on sipXecs for their communications needs Configuration Management ranging from 10 people offices with 4 analog trunk lines to companies True Plug & Play. Intuitive browser interface such as Amazon.com with thousands of users. for centralized control and management of dial plans, users and endpoints. sipXecs provides dramatic cost savings while delivering the system and User Self-control trunk redundancy, interoperability, scalability and plug–and-play benefits Powerful Web user portal puts the user in that SMB customers expect. sipXecs is a unified communications control to individually manage key features like time-based find-me / follow-me. solution based on a robust real-time architecture that grows with your business. Really easy to install and use sipXecs puts the user in control. It was designed so that everybody can be the sipXecs is the first business-grade IP PBX built on standards based open administrator. It installs in hours and all source. It offers proven, robust functionality for thousands of seats on an adds, moves and changes are done by the optionally redundant system, and provides low-cost communications user using a Web interface. solutions for main and remote offices, home workers, and call centers. Used Asterisk before? If you know Asterisk you will find sipXecs sipXecs interoperates with TDM PBXs via Voice over IP (VoIP) gateways, easy to use, much more stable, and enabling users to choose a phased-in approach to VoIP implementations. scalable. No more guessing about how things work, no more asking the experts. sipXecs relies on standards and utilizes proven Internet techniques and a The sipXecs project is adding new distributed architecture to create a highly secure and available IP voice capabilities fast following a disciplined system. roadmap. Switching to sipXecs has proven to be the right choice for many enterprises already. Open source offers lowest total cost of ownership http://www.sipfoundry.org (TCO) and it does not lock you into a vendor specific and proprietary solution. Overview Configuring the Dial Plan is easy sipXecs ECS Benefits Commercial support available as an option Pingtel’s Support and Maintenance services provide support and ongoing updates allowing you to realize return on investment (ROI) quickly. Easy to install, configure and manage Browser based system management tool allows you to Personal Auto-Attendant per user simply configure your server, managed devices (gateways and phones) and go! True intuitive management with click-and-go. Legacy telecommunications investment protection Standards-based system supports existing network and meets all requirements for TDM PBX replacement, augmentation and migration. Unmatched system flexibility, interoperability and lower-cost solution components SIP standards compliance ensures interoperability with off-the-shelf solution components and applications from other vendors that conform to SIP standards. Mix and match phones across the enterprise based on individual preferences. Elimination of expensive second phone lines for One – two – three clicks to configuring users, remote workers and additional trunk lines between phones and lines distributed offices Leverages existing broadband connections for voice and data. Enhanced employee productivity Delivers a wide range of innovative user features including Microsoft desktop and Exchange 2007 integration. Convenient VoIP system migration Begin with a single office and extend IP telephony to your remaining organization in a timeframe that suits you. Unified Messaging with Microsoft Outlook Future-proofed network 100-percent SIP standards-based system enables easy moves, adds and changes, installation of new features and deployment of new applications as they become available. Overview sipXecs ECS sipXecs Telephony System Features . User Management • Transfer (consultative & blind) • Create a user, provision a phone and assign a line in only • Call coverage three clicks – easy! • Call hold / retrieve • Numeric or alpha-numeric User ID • Consultation hold • User PIN management (UI or TUI) • Music on Hold for IETF standards compliant phones • Aliasing facility (numeric and alpha-numeric aliases) • Uploadable music file • Extension and alias uniqueness assurance • 3-way conference • Granular per user permissions • Call pickup (global and directed call pickup) • Call permissions: • Call park & retrieve o 900 Dialing • SIP URI dialing o International Dialing • CLID (Calling Line Identification) o Long Distance Dialing • CNIP (Calling party Name Identification Presentation) o Mobile Dialing • CLIP (Call Line Identification Presentation) o Local Dialing • CLIR (Call Line Identification Restriction) o Toll Free Dialing • Per gateway CLIP manipulation o Forward Calls External • Call waiting / retrieve • System permissions: • Do not Disturb (DnD) o User has voicemail inbox • Forward on busy, no answer, do not disturb o User listed in auto-attendant directory • Multiple line appearances o User can record system prompts • Multiple calls per line o User has superuser access • Multiple station appearance o User allowed to change PIN from TUI • Outbound call blocking • Custom permissions added by the administrator • Click-to-dial (Windows) • Supervisor permission for groups (e.g. Call Center • Redial supervisor) • Call history (dialed, missed, received) • SIP password management for security • Auto off-hook / ring down • User groups with group properties • Incoming only • Per user call forwarding (find me / follow me) • Configuration of individual Speed Dial softkeys o To local extension, PSTN number, or SIP • Auto-generation of Directory information address o Scheduled forwarding based on user defined User Self-Control individual schedules o Parallel or serial ring • Every user on the system gets access to a personal Web o Allows definition of ring time before trying next user portal for self-management and control number • Management of voicemail o Allows several forwarding destinations • Configuration of unified messaging preferences o Follow-me configuration using user portal • Time based find-me / follow-me • Extension pool with automatic assignment • Flexible configuration of call forwarding • Per user Caller ID (CLID) assignment • Personal call history • Per user Caller ID blocking • Personal phone book, speed dial and presence management PSTN Trunking • ACD presence and supervision capabilities • Individual phone management • Unlimited number of PSTN gateways and trunk lines • Direct Inward Dialing (DID) Superior Voice Quality • Local DID per gateway • DNIS • Peer-to-peer media routing with quality optimization • CLIP Management • Lower delay and jitter o User CLIP • Support for any codec supported by the phone o Gateway default CLIP • Codec negotiation directly between phones o Prefix stripping / appending • Per gateway CLIR • Automatic Route Selection (ARS) Dial Plan • Least-cost routing (LCR) • Easy to use GUI based dial plan manipulation • Automatic failover if unavailable • Time-based dialing rules with different admin defined • Automatic failover if busy schedules • FAX support (pass-through) • Rules based least cost routing • Mixing of PSTN trunks with SIP trunks • Automatic gateway redundancy and failover • Specific E911 routing • Permission based rules • Prefix manipulation • Dial plan templating for international dial plans Overview sipXecs ECS Performance Voicemail Subsystem • Unlimited number of simultaneous calls • Integrated voicemail system at no extra cost • Unlimited number of trunk lines • Browser based user portal for voicemail management • 54,000 BHCC, 120,000 BHCC redundant(dependant on • Message Waiting Indication (MWI) server platform) • User configurable distribution lists • Up to 10,000 users per dual-server HA system • Manage Notifications: • Automatic time distribution of re-registrations o Email notification of new voicemail messages o Forwarding of message as .wav file • Manage folders: Folders for message organization High Availability • Manage greetings: Multiple customizable greetings • Optionally fully redundant call control system • Operator
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