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June 22, 2006

Asterisk - Open Source PBX

Filed under: Mobile, Platform — Peter Kim @

Asterisk는 PBX 기본 및 그 이상의 기능을 Linux 혹은 MacOSX 상에서 구현한 Software이다. 많은 VoIP 프로토콜을 지원하고 있으며 업계 표준 규격을 모두 따르고 있다.

The Asterisk core module

  • PBX switch - 많은 유저와 자동 기능들 사이에서의 교환기 역할을 해주는 Core 모듈
  • Application Launcher - 음성사서함, 자동음악연주, 디렉토리 목록등 유저 부가 서비스 모듈
  • Codec Translator - 통신표준에 맞는 다양한 사운드 포맷/코덱 지원
  • Scheduler and I/O manager - 분산처리및 low-level에서의 scheduling

상용 제품에 버금가는 기능으로 호처리기능, CTI기능, 확장성, 코덱, 프로토콜, CTI호환, PRI호환에 대해서는 아래와 같다.

Call 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&trade ;)
  • 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
    • Flexible Mp3-based System
    • Random or Linear Play
    • Volume Control
  • 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
    • Visual Indicator for Message Waiting
    • Stutter Dialtone for Message Waiting
    • Voicemail to email
    • Voicemail Groups
    • Web Voicemail Interface
  • 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)
    • Allows direct connection of Asterisk PBX
    • 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

Codecs

  • ADPCM
  • G.711 (A-Law & μ-Law)
  • G.723.1 (pass through)
  • G.726
  • G.729 (through purchase of commercial license through Digium)
  • GSM
  • iLBC
  • Linear
  • LPC-10
  • Speex

Protocols

  • IAX™ (Inter-Asterisk Exchange)
  • H.323
  • SIP (Session Initiation Protocol)
  • MGCP (Media Gateway Control Protocol
  • SCCP (Cisco® Skinny&reg ;)

Traditional Telephony Interoperability

  • E&M
  • E&M Wink
  • Feature Group D
  • FXS
  • FXO
  • GR-303
  • Loopstart
  • Groundstart
  • Kewlstart
  • MF and DTMF support
  • Robbed-bit Signaling (RBS) Types
  • MFC-R2 (Not supported by Digium. However, a patch is available at Voip-info.org)

PRI Protocols

  • 4ESS
  • BRI (ISDN4Linux)
  • DMS100
  • EuroISDN
  • Lucent 5E
  • National ISDN2
  • NFAS

VoIP기능 및 특히 H.323, SIP, MGCP, SCCP 프로토콜 스택이 다 포함되어 있고, ADPCM, G.711, G.723.1, G.726, G.729, GSM, iLBC, Linear, LPC-10, Speex Codec을 다 지원한다. Open Source이니 해당 부분의 처리 로직도 확인할수 있다.

Asterisk : http://www.asterisk.org/

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