Digium AA50 Asterisk Appliance

Featuring the first Digium® developed AsteriskGUI, the Asterisk Appliance 50 brings small to medium enterprises a cost-effective, feature-rich, reliable telephony solution.

The Asterisk Appliance (AA50) is a standalone embedded Asterisk-based PBX targeted for small to medium businesses (2-50 users), remote branch offices of larger organizations (2-50 users per site), and managed service providers for on-premise CPE-based solutions with SIP or IAX trunking. The AA50 also offers a hybrid solution alternative (a combination of VoIP applications using legacy telecom equipment) for enterprise customers who are not yet ready to migrate to a complete VoIP phone system solution.

The Asterisk Appliance 50 features the commercially licensed Asterisk Business Edition software and the Digium developed AsteriskGUI. The AsteriskGUI makes the configuration, management, and fine tuning of your Asterisk system less complicated by providing an easy to use graphical interface. The AsteriskGUI is different from most interfaces that have been created for use with Asterisk, since changes in each interface are reflected in others in real time.

The AA50 is available in the following configurations: VoIP Only (S800i), Eight Trunk (S808B), and Four Station with Four Trunk (S844B).

Asterisk Appliance 50 Features

• Complete Asterisk Server with AsteriskGUI
• Embedded Asterisk Business Edition with commercial license
• Built-in Router Ideal for Small Offices
• Up to Eight Analog Ports
• 1GB Compact Flash® Card
• Hardware-based Echo Cancellation
• 8 MB Onboard Flash
• 64 MB Onboard RAM
• 5 Ethernet Ports (4 LAN, 1 WAN)

• All-in-one package of Digium Hardware, Digium Asterisk Software, system documentation, and subscription services.
• Increased reliability over PC-based systems
• Lower power requirements
• Hosted service providers can have a feature rich CPE-based solution with very short ROI
• Enables the deployment of fully featured cost effective telephony solutions with simple deployment features such as auto-provisioning of Polycom IP Phones
• Subscription support of any incidents, ongoing warranty, and access to the right people for peace of mind.

Asterisk Business Edition

Asterisk Business Edition is an enterprise-grade version of its acclaimed open source PBX for the Linux operating system.

Asterisk Business Edition provides tested reliability of critical functions and features. It tackles a wide range of challenges, from common PBX and key system replacements to highly-specialized applications. Asterisk Business Edition supports up to 40 simultaneous calls with upgrades to 240 calls available.

Why Asterisk Business Edition?
Asterisk is a complete telecommunications platform and represents a highly valuable software package for a number of reasons:

• Extreme cost reduction
• Control and customization
• Flexible dial plan
• Rich, broad feature base

Digium G.729 Codec

The industry standard G.729 codec allows more calls in limited bandwidth, utilizing IP voice in more cost effective ways. A typical VoIP call consumes 64Kbps of voice bandwidth. G.729 reduces the call to 8Kbps (normal IP overhead adds to this number). Asterisk currently supports G.729 Annex A only.

Digium provides an easy way for users to use G.729 compression and decompression for connecting to VoIP telephones and VoIP service providers with its G.729 codec for Asterisk.

High Performance
Echo Cancellation

For Asterisk users that connect to the PSTN, the most common type of echo is hybrid echo - the echo introduced by the impedance mismatch between 2-wire and 4-wire telephone circuits. The echo manifests as a distorted and delayed reflection of the users voice while in conversation with an external party through the PSTN.

Asterisk offers moderately effective echo cancellation routines that reduces the hybrid mismatch echo that most Asterisk users experience. However, there are cases in which these algorithms are not effective. To combat this, Digium introduced DSP-based echo cancellation modules for our multi-port T1/E1/J1 cards and our 24-port analog card.

Host-based Toll-Quality echo cancellation software operates under 32-bit Linux, providing echo cancellation for configurable tail lengths of 16ms (128 taps), 32ms (256 taps), 64ms (512 taps), and 128ms (1024 taps).

For new and existing customers of under-warranty Digium analog cards, this solution is offered with limited support, at no charge. This solution is also available, with no support, to customers of non-Digium Asterisk products at a per-channel rate.

LumenVox
Speech Engine

LumenVox Speech Engine offers up to 12,000 vocabulary items per recognition (depending upon the application) and includes the Speech Tuner.

LumenVox Speech Engine is a speech recognition engine that performs recognition on audio data from any audio source, and allows for dynamic language, grammar, audio format, and logging capabilities. It is also fully integrated with Asterisk.

The LumenVox Speech Engine is seamlessly integrated with the Asterisk platform and Dial Plan through the unique Asterisk Connector Bridge. Application developers can easily build speech-enabled IVR's by using the familiar Dial Plan scripting language or the C-API.

The LumenVox Speech Engine is speaker and hardware independent, supports industry standards such as SRGS and Semantic Interpretation, and performs better in challenging environments with highly efficient barge-in and noise reduction technology

The LumenVox Speech Engine is the core technology that handles the "recognition" of words and phrases. It is speaker-independent with no special training requirements. The engine relies on Grammars, which are a list of expected words or phrases that a caller might speak, to successfully recognize words.

Digium TE120P

The TE120P is a cost effective, high-performance digital telephony interface card that connects traditional telephony systems with emerging VoIP phone systems.

Used in conjunction with Asterisk®, the TE120P can be used to deliver a wide range of PBX and IVR services to the network or handset including Voicemail, Call Conferencing, Three-way Calling, and VoIP Gateways.

The TE120P is a selectable T1 (24-channel), E1 (32-channel), or J1 (24-channel) single span card. The card utilizes Digium's VoiceBus technology. VoiceBus technology allows the TE120P to use an industry standard bus-mastering PCI interface.

The TE120P supports both voice and data modes on its single span. For example, the card can support 12 channels dedicated to voice, routed directly to the Asterisk Open Source PBX, and 12 to data, handled by the underlying Linux operating system, thus eliminating the need for an external router. The TE120P works in both 3V and 5V slots by auto detecting the slot's voltage

By utilizing our TDMoE (TDM over Ethernet) technology, an exclusive Digium process, you can easily connect multiple PCs equipped with the TE120P and achieve voice quality on par with single PBX implementations. By adding multiple TE120Ps to each individual PC, you can easily scale this product, simple add additional cards as you need them for your expanding applications.

The TE120P supports industry standard telephony and data protocols, including both RBS and Primary Rate ISDN (PRI) protocol families for voice and PPP, Cisco HDLC, and Frame Relay data modes. The board drives both line-side and trunk-side interfaces, including call features.

Digium TE205P/TE207P
TE210P/TE212P

The TE205P and TE210P support both E1 and T1/J1 environments and are selectable on a per-card or per-port basis. This feature enables signaling translation between E1 and T1/J1 equipment and allows inexpensive T1/J1 channel banks to connect with E1 circuits. The TE205P and TE210P improves I/O speed over slave-only architecture, resulting in reduced CPU usage and increased card density per server.

Digium is fully compatible with existing software applications and it is fully integrated with Digium's Asterisk platform. The driver supports an API interface for custom application development and with the combination of Digium Hardware and Asterisk software, numerous combinations of telephony configurations are possible.

Providing termination for up to two T1, E1, or J1 circuits, up to 60 channels, the TE205P and TE210P support industry standard telephony and data protocols, including Primary Rate ISDN (both N. American and Standard Euro) protocol families for voice, PPP, Cisco, HDLC, and Frame Relay data modes. Both line-side and trunk-side interfaces are supported, also included are advanced call features.

The TE207P and TE212P are Digium's and the industry's first two-port digital cards featuring hardware-based echo cancellation. The new module enables users to eliminate echo tails up to 128ms or 1024 taps across all 64 channels in E1 mode or 48 channels in T1/J1 modes.

The TE205P and TE207P are intended for use only with a 5.0V PCI slot. The TE210P and TE212P are intended for use only with a 3.3V PCI slot.

Digium TE220

Digium's TE220 PCI Express card provides termination of up to 60 channels of voice or data across two E1, T1, or J1 interfaces. Supporting PCIe x1, the TE220 may be used in any available PCIe 1.0 compliant slot - 1x, x4, x8, and x16 without considerations for voltage selection or lane size. Selectable on a per-port or per-card basis, the TE220 allows E1 and T1 circuits to be mixed with full channel synchronization.

The TE220 may be combined with Digium's VPMOCT064 DSP-based echo cancellation module. The VPMOCT064 provides the G.168 algorithm performs 128ms (1024 taps) of echo cancellation across all 60 channels in E1 mode or all 48 channels in T1/J1 modes. Bundled with the VPMOCT064, the product SKU is TE220B.

Digium is fully compatible and integrated with Asterisk Business Edition, AsteriskNOW, and open source Asterisk.

The TE220 supports industry standard telephony protocols including North American and European Primary Rate signaling as well as standard Robbed Bit, Channel Associated Signaling in addition to standard PPP, HDLC, and Frame Relay data modes.

Digium TE405P/TE407P
TE410P/TE412P

The TE405P and TE410P support both E1 and T1/J1 environments and are selectable on a per-card or per-port basis, enabling signaling translation between E1 and T1/J1 equipment. The TE405P and TE410P improves I/O speed over slave-only architecture, which reduces CPU usage and increased card density per server.

Digium has designed the TE405P and TE410P to be fully compatible with existing software applications and it is fully integrated with Digium's Asterisk platform. Also, the driver supports an API interface for custom application development. With the combination of Digium Hardware and Asterisk software, numerous combinations of telephony configurations are possible.

Providing termination for up to four T1, E1, or J1 circuits, up to 120 channels, the TE405P and TE410P support industry standard telephony and data protocols, including Primary Rate ISDN (both N. American and Standard Euro) protocol families for voice, PPP, Cisco, HDLC, and Frame Relay data modes. Included are advanced call features with line-side and trunk-side interface support.

The TE407P and TE412P are bundlings of our leading TE405P and TE410P products and our new VPMOCT128 G.168-compliant echo cancellation module. The TE407P and TE412P are the industry's first two-port digital cards featuring hardware-based echo cancellation. The new module eliminates echo tails up to 128ms or 1024 taps across all 120 channels in E1 mode or 96 channels in T1/J1 modes.

The TE405P and TE407P are intended for use only with a 5.0V PCI slot. The TE410P and TE412P are intended for use only with a 3.3V PCI slot.

Digium TE420

Digium's TE420 PCI Express card provides termination of up to 120 channels of voice or data across four E1, T1, or J1 interfaces in a PCIe x1 form factor. The TE420 allows E1 and T1 circuits to be mixed with full channel synchronization. The TE420 may be used in any available PCIe 1.0 compliant slot - 1x, x4, x8, and x16 regardless of voltage selection or lane size.

The TE420 may be combined with Digium's VPMOCT128 DSP-based echo cancellation module. The VPMOCT128 provides the G.168 algorithm which has been labeled a benchmark for echo cancellation and performs 128ms (1024 taps) of echo cancellation across all 96 channels in T1/J1 modes or all 128 channels in E1 mode.

Digium has designed the TE420 to be fully compatible and integrated with Asterisk Business Edition, AsteriskNOW, and open source Asterisk.

The TE420 supports industry standard telephony protocols including North American and European Primary Rate signaling as well as standard Robbed Bit, Channel Associated Signaling in addition to standard PPP, HDLC, and Frame Relay data modes.

Digium B410P

Using a Linux PC, Digium's digital hardware, and Asterisk software, companies can create an intelligent telephony environment which has every sophisticated feature of a high-end PBX/Voicemail platform.

The B410P is a half-length, full-height universal 3.3V and 5.0V 32-bit PCI 2.2 card supporting four BRI S/T interfaces. Each of the four ports of the B410P can be independently configured for TE or NT mode. The card's on-board hardware echo cancellation performs at 64ms or 512 taps per channel for each of the eight voice channels.

Digium TDM400P

The TDM400P is a PCI 2.2 compliant telephony card that connects analog telephones and analog POTS lines through a PC card. The card supports Station (FXS) and Trunk (FXO) interfaces.

Using Digium's Asterisk PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system.

The TDM400P brings the telephony system price point to a low level by taking the place of an expensive channel bank. Using Digium's single Station (S110M) and Trunk (X100M) modules with the TDM400P, companies can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM400P cards populated with modules.

Digium TDM800P

The TDM800P is a PCI 2.2-compliant telephony card for connecting analog telephones and analog POTS lines through a PC. It supports combinations of Station (FXS) and/or Trunk (FXO) modules for a total of 8 lines.

The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. The TDM800P reduces part complexity, cable clutter, and points of failure by effectively eliminating the need for multiple brackets, external dongles, or splitters.

The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® Quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports.

The optional hardware echo cancellation module provides superior voice quality on both Trunk and Station interfaces. It provides 1024 taps (128 milliseconds) of G.168 compliant echo cancellation.

Digium AEX800

The AEX800 is a PCI-Express x1 1.0-compliant telephony card for connecting analog telephones and analog POTS lines through a PC. It supports combinations of Station (FXS) and/or Trunk (FXO) modules for a total of 8 lines.

As with the TDM800P, the AEX800 provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket, the AEX800 reduces part complexity, cable clutter, and points of failure while providing a more scaleable solution.

The AEX800 contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® Quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports.

The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of G.168 compliant echo cancellation for superior voice quality on both Trunk and Station interfaces.

Digium TDM2400P

The TDM2400P is a full-length modular twenty-four port PCI 2.2-compliant telephony card for connecting analog telephones and POTS lines through a PC. It supports a combination of up to 6 Station (FXS) and/or Trunk (FXO) modules for a total of 24 lines.

Using an industry-standard bursting, bus-mastering PCI interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM2400P eliminates the requirement for separate channel bank and T1 interface cards, at an industry-leading price. The Quad Station (S400M) and Quad Trunk (X400M) modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module (VPMADT032) provides superior voice quality: 1024 taps (128 milliseconds) of G.168 compliant echo cancellation on both Trunk and Station interfaces. Scaling of this solution is accomplished by adding additional TDM2400P cards.

Digium S101i - "IAXy"

The Digium® S101I, affectionately known as the IAXy, takes Asterisk® from the PC to the CPE. The IAXy provides a single, fully featured Station interface with an Ethernet back-end, speaking the Asterisk-native IAX protocol, at a highly competitive price. IAXy can robustly transfer calls between endpoints, allowing on-net calls to be moved off of a service provider's network for better quality and lower cost. The IAXy is aimed at Voice Over Broadband and Internet Telephone Service Providers. The IAX protocol provides complete NAT transparency, enabling full operation behind NAT and PAT firewalls.

The IAXy features include: Auto Upgrade, Caller ID, Call Waiting, Cancel Call Waiting, Caller ID Disable / Enable, Three-way Calling, Caller ID on Call Waiting, Blind Transfer, Call Parking, Voice Mail Waiting Indicator, Mute Rx on-Hook, Pulse Dial, and Call Hold. It supports the G.711 u-law and ADPCM voice codecs.

Digium PWR2400B

The PWR2400B is a device accessory that supplies additional power to Digium analog cards mounted with Station modules when no internal power cables are available from the PC or server.

Station modules provide ringing voltage and battery to telephones or handsets and require substantial amount of 12V DC power in order to operate properly. The PCI interface cannot provide sufficient voltage necessary to ring 4, 8 or 24 modules. Digium analog cards are mounted with a 4-pin disk-drive power connector on the base board to supplement the card power. This connector must be supplied with an internal power connector from the system's power supply if Station modules are in use.

For systems without available internal 12V power connectors, the PWR2400B is the solution. The PWR2400B provides the 12V DC power to analog interface cards that is necessary to drive Station modules. Trunk modules do not require any external power resources.

Digium TC400B

The TC400B is a PCI 2.2-compliant card that transforms complex VoIP codecs into simple codecs.

The TC400B is a half-length, low-profile PCI-2.2 compliant TC400P base card bundled with the TC400M voice processing module. The TC400B is designed to handle, in dedicated DSP resources, the complex codec translations for highly compressed audio that would otherwise be processed by Asterisk in software.

Asterisk, in software and with Digium G.729a licensing, is capable of transforming the G.729a codec into other codecs for the purposes of call origination or termination, bridging disparate calls, or VoIP to TDM connectivity. These transformations in software are very expensive, in terms of MIPS, and require a substantial amount of CPU time to accomplish. The TC400B relieves the CPU of this duty, freeing it up to handle other tasks or to complete additional call processing; and also provides Asterisk with the capability of bridging G.723.1 compressed audio into other formats.

The TC400B decompresses G.729a (8.0kbit) or G.723.1 (5.3kbit) into u-law or a-law; or, compresses u-law or a-law into G.729a (8.0kbit) or G.723.1 (5.3kbit). The TC400B is rated to handle up to 96 bi-directional G.729a transformations or 92 bi-directional G.723.1 transformations. The TC400B does not require additional licensing fees for the use of these codecs.