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Custom PBX Solutions Print

Twin Geckos provides asterisk based PBX solutions for all sized businesess.
We'll evaluate your needs and set up a system that will work for you!

What Is Asterisk?

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 three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.

Asterisk needs no additional hardware for Voice over IP.  For interconnection with digital and analog telephony equipment, Asterisk supports a number of hardware devices, most notably all of the hardware manufactured by Asterisk's sponsors, Digium™. Digium has 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.

Also supported are the Internet Line Jack and Internet Phone Jack products from Quicknet.

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.

 

Features at a glance: 

 

  • Add or change extension and voicemail accounts in seconds
  • Supports SIP, IAX, and ZAP clients
  • Supports VoIP and ZAP trunks
  • Reduce long distance costs with LCR
  • Route incoming calls based on time-of-day
  • Create interactive Digital Receptionist menus
  • Design sophisticated call groups
  • Manage callers with Queues
  • Upload custom on-hold music (MOH)
  • Search company directory, based on first or last name
  • Detect and receive incoming faxes
  • Share administrative duties with AMP Users
  • View call detail reporting with asterisk-stat
  • View extension and trunk status with Flash Operator Panel

  • Introduction—What is VoIP?

    Voice over IP or VoIP is a term used in IP telephony for a
    set of facilities that use the Internet Protocol (IP) to deliver
    voice information. In general, this means sending voice
    information in digital form in discrete packets rather than
    in the traditional circuit-committed protocols of the public
    switched telephone network (PSTN). A major advantage
    of VoIP and Internet telephony is that they avoid the tolls
    charged by ordinary telephone service.

    The term VoIP derives from the VoIP Forum, an effort by
    major equipment providers, to promote the use of ITU-T
    H.323, the standard for sending voice (audio) and
    video using IP on the public Internet and within an
    intranet. The Forum also promotes the user of directory
    service standards so that users can locate other users
    and the use of touch-tone signals for automatic call distribution
    and voice mail.

    In addition to IP, VoIP uses the real-time protocol (RTP) to
    ensure that packets get delivered in a timely way.
    Because the nature of public networks such as the
    Internet makes it difficult to guarantee Quality of Service
    (QoS), better service is usually possible with private networks
    managed by an enterprise or by an Internet
    telephony service provider (ITSP).

    A technique used by at least one equipment manufacturer,
    Adir Technologies (formerly Netspeak), to help
    ensure faster packet delivery is to use the ping utility to
    contact all possible network gateway computers that
    have access to the public network and choose the
    fastest path before establishing a Transmission Control
    Protocol (TCP) sockets connection with the other end.
    Enterprises use VoIP gateways to enter into the VoIP
    environment. A gateway receives packetized voice
    transmissions from users within the company and then
    routes them to other parts of the company’s intranet
    (local area or wide area network) or—using a T-carrier
    system or E-carrier interface—sends them over the public
    switched telephone network.
     

    VoIP Standards

    • ITU-T H.320 Standards for Video Conferencing
    • H.323 ITU Standards
    • H.324 ITU Standards
    • VPIM Technical Specification

    VOIP Glossary

    Capabilities 

    Asterisk provides for standard CLASS features such as three-way calling, Caller ID, call return, call waiting, call-transfer-disconnect, as well as providing advanced features such as Voicemail services with Directory and Email Delivery, Conferencing, Interactive Voice Response, Call Queuing, Paging, Auto Attendant, Music on Hold, Voice Recognition and Voice Synthesis, and more.

    Asterisk can also act as a bridge or gateway between traditional TDM/PSTN networks and Voice over IP networks. Asterisk speaks SIP, H.323, MGCP, and IAX on the Voice over IP side, and can also speak PRI, Robbed Bit Signalling [FXS/FXO, Loopstart, Groundstart, Kewlstart, E&M, E&M Wink, FGD], and GR303.

     What Can you do with Asterisk?

    Asterisk is more than just a PBX, it's a telephony tool-kit. There is no boundary to what one can create with Asterisk, but here are a few ways you can make use of this software:

    • Hosted PBX Services -- Sell a "Virtual" PBX system to customers.
    • Voicemail System -- Provide advanced voicemail services with follow-me, email delivery, and more.
    • Long Distance Platform -- Asterisk can be used as a long distance platform to provide long-distance services to customers using a toll-free "dial-around" access number. CLECs use Asterisk to provide long distance services to their resale end-users, bypassing the Bell PIC process.
    • Calling Card Platform -- Asterisk can be used as a calling card platform. Voice over IP Gateway -- You can use Asterisk as a gateway between VoIP providers or VoIP and the Public Switched Telephone Network.
    • Conference Calling Platform -- Sell conference calling with advanced features.

    Asterisk can be used in a purely VoIP environment, but it is often useful to connect Asterisk to the Public Switched Telephone Network.

    Digium is a company that offers inexpensive and feature-rich PSTN interface cards that work natively with Asterisk. These interface cards are compatible with commodity server equipment. Digium offers cards with FXO and FXS analog ports, single T1/E1 interfaces, and quad T1/E1 interfaces. Using the Digium interface cards, you can use PRIs, Channelized T1s, or Feature Group D trunks to connect your Asterisk box to the telephone network.

    Another option is to buy service from a VoIP service provider. TxLink offers extremely competitive rates for VoIP termination (long distance) and origination (DID phone numbers) that allow you to hook up your Asterisk box to the public network without having to buy any hardware or install any TDM trunks. Click here to open a new window for more information on TxLink's VoIP service offerings. TxLink offers IAX connectivity so you can connect your Asterisk box directly into TxLink's network.

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