digium asterisk

Digium™ is the creator and primary developer of Asterisk™, the industry’s first Open Source PBX. Used in combination with Digium’s PCI telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over TDM, switched, IP, and Ethernet architectures. Digium solutions reduce the costs of traditional TDM and VoIP implementations through open source, standardsbased software and innovative hardware solutions, including legacy PBX, IVR, auto-attendant, and next-generation gateways, media servers, and application servers. Digium hardware supports traditional voice protocols, including PRI, RBS, FXS, FXO, E&M, Feature Group D, Groundstart, and Loopstart. Data protocols include PPP, Cisco HDLC, and Frame Relay. For packet voice, Asterisk supports IAX (Inter-Asterisk Exchange), SIP, MGCP, Skinny, and H.323 VoIP protocols. Digium provides a highly refined selection of quality hardware and software products, developed and implemented using innovative engineering techniques (primarily open source development). A full range of professional services complement these product lines, including consulting, technical support, and custom software development services. The open source communications revolution is here, and Digium is leading the way.

About Digium

Digium™ is the creator and primary developer of Asterisk™, the industry’s first Open Source PBX.

Used in combination with Digium’s PCI telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over TDM, switched, IP, and Ethernet architectures.

Digium solutions reduce the costs of traditional TDM and VoIP implementations through open source, standardsbased software and innovative hardware solutions, including legacy PBX, IVR, auto-attendant, and next-generation gateways, media servers, and application servers.

Digium hardware supports traditional voice protocols, including PRI, RBS, FXS, FXO, E&M, Feature Group D, Groundstart, and Loopstart. Data protocols include PPP, Cisco HDLC, and Frame Relay. For packet voice, Asterisk supports IAX (Inter-Asterisk Exchange), SIP, MGCP, Skinny, and H.323 VoIP protocols.

Digium provides a highly refined selection of quality hardware and software products, developed and implemented using innovative engineering techniques (primarily open source development). A full range of professional services complement these product lines, including consulting, technical support, and custom software development services.
The open source communications revolution is here, and Digium is leading the way.

Applications

PBX
Asterisk allows you to create a PBX that rivals the features and functionality of traditional telephony switches. Other PBXs are expensive, proprietary, and now passé. Asterisk is cost-effective, low-maintenance, and flexible enough to handle all voice and data networking. With Asterisk, Digium hardware, and a common PC, anyone can replace an existing switch or complement a PBX by adding VoIP, voicemail, conferencing and many other capabilities. Asterisk integrates with most standards- based IP telephone handsets and software. Analog phones and ADSI screen phones are also supported.

Proven through installations around the globe, Asterisk is solid and stable in SOHO or Enterprise environments.

Interactive Voice Response (IVR)

Asterisk’s flexible IVR capability allows a user to interact with a database using a menu of pre-recorded voice-clips.

Using MySQL and other popular databases, Asterisk can interact with the caller through touch tone inputs, record responses, query databases, and utilize AGI scripts to perform specific tasks. For example, a customer can authenticate a pre-paid calling card with a PIN queried from a database. The Asterisk IVR will give the number of remaining minutes and later disconnect if that customer runs out of time. The spooling feature can allow Asterisk to dial a list of numbers from a database to give warnings during homeland security emergencies. Asterisk IVR allows developers to create a myriad of IVR solutions.

Auto-Attendant
Asterisk’s auto-attendant features include greetings, extended greetings, music-on-hold, voice message forwarding and message appending. Asterisk plays music or pre-recorded messages to customers on hold.

Music can be sorted into various folders. Separate auto-attendant feature sets can be used for different situations. The voicemail tree supports directories by department, employee, extension, etc., offering flexibilityand allowing a small company to appear large. Unbound by the limits of traditional voicemail, Asterisk can support an unlimited number of simultaneous ports.

Conference Bridge
The Meet Me Bridge is fully integrated into Asterisk and supports features essential for business conferences, saving the Asterisk user from what was once a huge expense. The Conference Chairperson can select a “listen only” or a “talk and listen” conference. When the Chairperson hangs up the other parties are disconnected. Conferences may be securely accessed only through a pre-defined PIN.

Many companies have also created chat services based on Asterisk Meet Me, with many chat rooms that users can transfer between.

Media Server
Asterisk augments existing PBXs and Gateways with select features for either PSTN or IP protocols. Acting as an adjunct to a legacy system or soft switch, Asterisk can extend features and functionality by providing voicemail and conferencing services.

Asterisk can also retrofit traditional TDM PBXs with VoIP extensions to remote offices which appear as normal extensions of the PBX.

VoIP and Protocol Gateway
Asterisk’s broad support of both traditional TDM and VoIP protocols permits the construction of flexible gateways between different channel types. Using Asterisk, it is not only easy to create many common varieties of protocol converters, translating between T1, E1, PRI, SIP, IAX, GR-303, MGCP, FXS, and many others. It enables you to create more sophisticated gateways and gateways with redundant links. For example, an MGCP to SIP gateway with a PRI backup can be created in case SIP trunks are unavailable— the possibilities are nearly endless.

VoIP Switch
Asterisk can act as a soft switch in addition to acting as a traditional TDM switch, allowing it to control a variety of devices including phones, gateways, media servers, and other Asterisk servers. It can handle virtually any VoIP protocol, including SIP, IAX, H.323, MGCP, and  Skinny.

Asterisk collects call detail records and provides a variety of billing options (including Open Settlement Protocol) and may be configured to carry media (especially useful for SIP+NAT situations) or to have devices send media directly to one another. Asterisk adds extra IP or PSTN capabilities to existing PBXs and Gateways. Asterisk extends features and functionality by providing voicemail and conferencing services when acting as an adjunct to a legacy system or soft switch.

Asterisk can also add remote VoIP office extensions to traditional TDM PBXs which appear as normal extensions from the preexisting PBX.