About the Active-Sites Asterisk based VoIP Phone System This interface is the combined effort of web developers, software engineers, web hosting providers, security solutions providers, technology consultants, and network administrators from Active-Sites, Inc, and GSI Systems, Inc. Special thanks to Mark Spencer of Digium for creating Asterisk. Asterisk replaces traditional PBX hardware with software. We integrate the most desirable features into a cohesive web interface driven system. We can help you lower your telecommunications costs while providing more functionality in a user friendly package. Why you should consider an Asterisk based solution ? ADDITION FEATURES FOR LESS Proprietary VoIP phone systems often utilize central servers costings upwards of $45,000. An Asterisk * VoIP based system can be built and installed for thousands less and includes options such as voicemail to email as well as many other features that are sold as add-on modules by proprietary VoIP server vendors. COMPATIBILITY The SIP protocol is a standard IP protocol for voice communications. An analogy may be made to the HTTP protocol that is utilized for web browsing. These protocols are cross-platform and non-hardware specific. The Asterisk server handles the SIP protocol as well as other standard and proprietary protocols. The benefit of utilizing standards based protocols for both the server and the phones is that your choice of hardware is not limited to one vendor. SCALABILITY AND INTEROPERABILITY The system allows for almost unlimited expansion. Non-software based systems have limits on the number of extensions, whereas software based systems are only limited by the available network bandwidth and the server horsepower, both of which can be increased relatively inexpensively with commodity network hardware. TOTAL COST OF OWNERSHIP Most non-software based systems also have a much higher total cost of ownership, as they require separate wiring and technician visits to modify the physical wiring to the PBX whenever an owner of an extension moves offices, or other system changes are required. The SIP based software solution allows the extension configuration to follow the phone and thus office moves are a trivial matter. Configuration changes are performed in the software realm and thus do not require a technician visit. A software based solution will never reach maximum extension capacity and thus will never require replacement due to lack of system extensions. You simply expand to meet your needs. Communication can be routed over the least cost route and the voice often rides for free if a data connection is already in place between the two endpoints, based on a software configurable options. Routable options include private T1 connections , the internet, POTS lines, ISDN lines, or LAN connections. The lack of yearly license and upgrade fees also lowers the total cost of ownership. RELIABILTY Asterisk * PBX runs on Linux rather than the other proprietary operating system that most other VoIP vendors utilize and thus is far less prone to suffer downtime due to viruses and worms. Most facilities have a trained IT staff to administer their data network and rely on outside vendors for their voice network. The data and voice are able to utilize the same cabling infrastructure and thus can be administered by existing IT staff. Asterisk * PBX Features Telephony Services * Voicemail System * Password Protected * Separate Away and Unavailable Messages * Default or Custom Messages * Multiple Mail Folders * Web Interface for Voicemail Checking * E-mail notification of Voicemail with .WAV attachment * Voicemail Forwarding * Visual Message Waiting Indicator * Message Waiting Stutter Dialtone * Auto Attendant * Interactive Voice Response * Overhead Paging * Flexible Extension Logic * Multiple Line Extensions * Multi-Layered Access Control * Direct Inward System Access * Directory Listing * Conference Bridging * Unlimited Conference Rooms * Access Control * Call Queuing * ADSI Menu System * Support for Advanced Telephony Features * PBX Driven Visual Menu Systems * Visual Notification of Voicemail * Call Detail Records with Web based reporting * Local Call Agents * Remote Call Agents * Protocol Bridging * Provides seamless integration of technologies * Offers a unified set of services to users regardless of connection type * Allows interoperability of VoIP systems Call Features * Music on Hold * Music on Transfer * Flexible mp3 based system * Volume Control * Random Play * Linear Play * Call Waiting * Caller ID * Caller ID Blocking * Caller ID on Call Waiting * Call Forward on Busy * Call Forward on No Answer * Call Forward Variable * Call Transfer * Call Parking * Call Retrieval * Remote Call Pickup * Do Not Disturb Scalability * TDMoE * Allows Direct Connection of Asterisk PBX * Offers Zero Latency * Uses Commodity Ethernet Hardware * Voice over IP * Allows for Integration of Physically Separate Installations * Uses commonly deployed data connections * Allows a unified dialplan across multiple offices Voice over IP Interoperability: Asterisk™ provides transparent bridging between Voice over IP protocols and traditional telephony equipment.  In addition, Asterisk can transfer calls from one system to another via the Inter-Asterisk™ eXchange (IAX™) protocol. * Inter-Asterisk™ eXchange (IAX™) * H.323 * Session Initiation Protocol (SIP) * Media Gateway Control Protocol (MGCP) Traditional Telephony Interoperability * Robbed Bit Signaling Types * FXS and FXO * Loopstart * Groundstart * Kewlstart * E&M * E&M Wink * Feature Group D * PRI Protocols * 4ESS * Lucent 5E * DMS100 * National ISDN2 * EuroISDN * BRI (ISDN4Linux) Codec Support * GSM * G.729 (available through purchase of commercial license(s)) * G.723.1 (pass through) * Linear * Mu-Law * A-Law * ADPCM * G.726 * ILBC * LPC-10 * MP3 (decode only) Fax Support * Fax Forwarding to an extension.