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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. Back
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