| Asterisk |
|
|
|
|
The Open Source Linux PBX  Asterisk is a complete IP telecommunications platform. It functions like a traditional PABX but it has added advantages due to its IP Protocol structure. From caller ID to long distance, anything your telephone system can do, Asterisk can do better—and cheaper. Asterisk was designed to be able to do everything a traditional telephone system can do, and much, much more. Asterisk is, at it's heart, a PBX system. However, it includes a whole host of telephony features such as voicemail and call conferencing. Asterisk creates a PBX that rivals the features and functionality of the latest top-end telephony switches. Asterisk is cost-effective, low-maintenance, and flexible enough to handle all voice and data networking. Proven through installations around the globe, and connecting the corner office to the most remote company outpost, Asterisk is solid and stable in either small office or Enterprise environments. Asterisk greatly reduces the cost of traditional telecommunication technology and operation, and moves voice over IP, VoIP, to the mainstream. Asterisk integrates a pre-existing analog telephone network with the benefits of IP technology, greatly reducing costs. Asterisk-based telephony solutions offer a rich and flexible feature set. Asterisk offers both classical PBX functionality and advanced features, and interoperates with traditional standards-based telephony systems and Voice over IP systems. Asterisk offers the features one would expect of a large proprietary PBX system such as Voicemail, Conference Bridging, Call Queuing, and Call Detail Records. Asterisk is a complete PBX in software. It runs on Linux and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in many protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.  Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response and Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway). Check the Features section for a more complete list. Asterisk needs no additional hardware for Voice over IP. For interconnection with digital and analog telephony equipment, Asterisk supports a number of hardware devices such as single and quad span T1 and E1 interfaces for interconnection to PRI lines and channel banks as well as a single port FXO card and a one to four-port modular FXS and FXO card. Asterisk supports a wide range of TDM protocols for the handling and transmission of voice over traditional telephony interfaces. Asterisk supports US and European standard signalling types used in standard business phone systems, allowing it to bridge between next generation voice-data integrated networks and existing infrastructure. Asterisk not only supports traditional phone equipment, it enhances them with additional capabilities. Using the Inter-Asterisk eXchange (IAX™) Voice over IP protocol, Asterisk merges voice and data traffic seamlessly across disparate networks. While using Packet Voice, it is possible to send data such as URL information and images in-line with voice traffic, allowing advanced integration of information. Asterisk provides a central switching core, with four APIs for modular loading of telephony applications, hardware interfaces, file format handling, and codecs. It allows for transparent switching between all supported interfaces, allowing it to tie together a diverse mixture of telephony systems into a single switching network. Asterisk-based telephony solutions offer a rich and flexible feature set. Asterisk offers both classical PBX functionality and advanced features, and interoperates with traditional standards-based telephony systems and Voice over IP systems. Asterisk offers the features one would expect of a large proprietary PBX system such as Voicemail, Conference Bridging, Call Queuing, and Call Detail Records. Asterisk™ Features • ADSI On-Screen Menu System • Alarm Receiver • Append Message • Authentication • Automated Attendant • Blacklists • Blind Transfer • Call Detail Records • Call Forward on Busy • Call Forward on No Answer • Call Forward Variable • Call Monitoring • Call Parking • Call Queuing • Call Recording • Call Retrieval • Call Routing (DID & ANI) • Call Snooping • Call Transfer • Call Waiting • Caller ID • Caller ID Blocking • Caller ID on Call Waiting • Calling Cards • Conference Bridging • Database Store / Retrieve • Database Integration • Dial by Name • Direct Inward System Access • Distinctive Ring • Distributed Universal Number Discovery (DUNDi™) • Do Not Disturb • E911 • ENUM • Fax Transmit and Receive (3rd Party OSS Package) • Flexible Extension Logic • Interactive Directory Listing • Interactive Voice Response (IVR) • Local and Remote Call Agents • Macros • Music On Hold • Music On Transfer o Flexible Mp3-based System• Predictive Dialer • Privacy • Open Settlement Protocol (OSP) • Overhead Paging • Protocol Conversion • Remote Call Pickup • Remote Office Support • Roaming Extensions • Route by Caller ID • SMS Messaging • Spell / Say • Streaming Media Access • Supervised Transfer • Talk Detection • Text-to-Speech (via Festival) • Three-way Calling • Time and Date • Transcoding • Trunking • VoIP Gateways • Voicemail o Visual Indicator for Message Waiting• Zapateller Computer-Telephony Integration  • AGI (Asterisk Gateway Interface) • Graphical Call Manager • Outbound Call Spooling • Predictive Dialer • TCP/IP Management Interface Scalability • TDMoE (Time Division Multiplex over Ethernet) o Allows direct connection of Asterisk PBX• Voice-over IP o Allows for integration of physically separate installationsKey Specifications/Special Features: • SIP 2.0 RFC3261 • Touch-tone (DTMF) • RFC2833 (DTMF) • Multi lines; Multiple proxies - available after Q1, 2005 • Call hold; call transfer; call forward; call timer; mute; speed dail • Dial/redial/hang up • Dynamic CODEC selection • Call ID (SIP ID) • Missed calls indication • Recent 60 calls dialed; recent 60 calls received • Direct IP to IP calling • Phonebook, 500 contacts with 2 numbers per contact • 5 personalize music tones • Clock, manual or SNTP • 3-way conference; local voice message; record; backlight - available after Q1, 2005 • Data/voice/protocol encryption - depends on the ISPs • 4 x 20 Char-based LCD or 112 x 56 pixel-based LCD -available after Q1, 2005 • G.723.1/G.729/G.711 Codec • G.722 Codec - available after Q1, 2005 • AEG G. 168 Key Specifications/Special Features: • SIP 2.0 RFC3261 • Touch-tone (DTMF) • RFC2833 (DTMF) • Multi lines; multiple proxies - available after Q1, 2005 • Call hold; call transfer; call forward; call timer; mute; speed dail • Dial/redial/hang up • Dynamic CODEC selection • Call ID (SIP ID) • Missed calls indication • Recent 60 calls dialed; recent 60 calls received • Direct IP to IP calling • Phonebook, 500 contacts with 2 numbers per contact • 5 personalize music tones • Clock, manual or SNTP • 3-way conference; local voice message; record; backlight - available after Q1, 2005 • Data/voice/protocol encryption - depends on the ISPs • 4 x 20 char-based LCD or 112 x 56 pixel-based LCD -available after Q1, 2005 • G.723.1/G.729/G.711 Codec • G.722 Codec - available after Q1, 2005 • AEG G. 168 |
| Next > |
|---|