Introducing Asterisk

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Justo odio dignissim qui blandit praesent luptatum zzril delenit augue duis dolore te feugait lorem ipsum dolo. Lorem ipsum dolorsit amet, conse ctetuer adipiscing elit, sed diam nonummy nibh euismod elit adispe. Feugait nulla facilisi. Lorem ipsum dolorsit amet, conse ctetuer laoreet dolore magna aliquam erat volutpat. Magna aliquam erat volu isi enim ad minim veniam melo eratta dolore magnus. Information Technology Asterisk is the product of ten years of work by a com- munity of thousands from around the globe. The com- munity is made up of users, developers and advo- cates who have contributed their time and efforts to make Asterisk what it is today. Asterisk is an Open Source PBX that provides the same functionality as high-end business telephone systems. It is the most flexible and scalable telephone system on the market, providing a broad array of features that are not yet available in even the most advanced pro- prietary systems. The software is free and runs on inexpensive Linux servers, which also makes it the cheapest telephone system on the market today! Asterisk Consulting provides solutions to improve organisational efficiency by implementing man- aged information and communication systems. We offer expert support and advice, throughout Middle East and Gulf region. The Asterisk Telephone System has been fully implemented by Asterisk Consulting in a number of state and private organisations, offering su- perb functionalities, with no licensing costs. We have a full team of in-house accredited dCAP (Digium Certified Asterisk Professional) and SIP Master engineers! Asterisk™ is the world’s most popular and powerful open source telephony platform. With Asterisk virtually any com- puter can become a telephony application server, drastically reducing the cost of ownership and delivering a noticeable return on investment that no proprietary solution can match. Open source Asterisk works for SMBs, Corporate Enterprises, Government and Contact Cen- ters, offering robustness, ad- vanced features and absolute flexibility and scalability. Get the most from your Asterisk solution ASTERISK Open Source Telephony

description

Introducing Asterisk with it's features

Transcript of Introducing Asterisk

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luptatum zzril delenit augue duis dolore te

feugait lorem ipsum dolo.

Lorem ipsum dolorsit amet, conse ctetuer

adipiscing elit, sed diam nonummy nibh

euismod elit adispe.

Feugait nulla facilisi. Lorem ipsum dolorsit

amet, conse ctetuer laoreet dolore magna

aliquam erat volutpat.

Magna aliquam erat volu isi enim ad minim

veniam melo eratta dolore magnus.

Information Technology

Asterisk is the product of ten years of work by a com-

munity of thousands from around the globe. The com-

munity is made up of users, developers and advo-

cates who have contributed their time and efforts to

make Asterisk what it is today.

Asterisk is an Open Source PBX that provides the

same functionality as high-end business telephone

systems.

It is the most flexible and scalable telephone system

on the market, providing a broad array of features that

are not yet available in even the most advanced pro-

prietary systems.

The software is free and runs on inexpensive Linux

servers, which also makes it the cheapest telephone

system on the market today!

Asterisk Consulting provides solutions to improve

organisational efficiency by implementing man-

aged information and communication systems.

We offer expert support and advice, throughout

Middle East and Gulf region.

The Asterisk Telephone System has been fully

implemented by Asterisk Consulting in a number

of state and private organisations, offering su-

perb functionalities, with no licensing costs.

We have a full team of in-house accredited dCAP

(Digium Certified Asterisk Professional) and SIP

Master engineers!

As te r i sk™ is the wor ld ’s

mos t popu la r and power fu l

open sou rce te l ephony

p l a t fo rm.

With Asterisk virtually any com-

puter can become a telephony

application server, drastically

reducing the cost of ownership

and delivering a noticeable return

on investment that no proprietary

solution can match.

Open source Asterisk works for

SMBs, Corporate Enterprises,

Government and Contact Cen-

ters, offering robustness, ad-

vanced features and absolute

flexibility and scalability.

Get the m os t f rom your

As ter i sk so lu t ion

ASTERISK Open Source Telephony

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Asterisk offers an Open Source, cross-platform Instant Messag-

ing client optimized for businesses and organizations. It features

built-in support for group chat, telephony integration, and strong

security. It also offers a great end-user experience with features

like presence, group chat and tabbed conversations. It is written

in Java and can be used standalone or as an add-on or plugin to

certain Web browsers. Combined with the Openfire server,

Spark IM is the easiest and best alternative to using unsecure

public Instant Messaging networks.

Openfire supports the following features:

♦ Web-based administration panel

♦ Plugin interface

♦ Customizable

♦ LDAP connectivity

♦ User-friendly web interface and guided installation

♦ Database connectivity to store messages and contacts

F E A T U R E S

∗ Voicemail to email

∗ MS Outlook integration

∗ Auto Attendant

∗ Call recording

∗ Name dialing

∗ Call reporting

∗ Music/Message on hold

∗ Automatic Route Selection

∗ Remote Extensions

∗ Call Distribution

∗ Flexibility and scalability

B E N E F I T S

∗ No Licensing costs

∗ Significant call savings

∗ Improved SLA levels

∗ Enhanced productivity

∗ Open Source platform

∗ Supports the use of exist-

ing servers to comply with

EU recycling guidelines

Jo in the As te r is k Open

Source Commun i ty

The main scope of any organisation choosing to migrate from their

existing system to a full Asterisk solution rests within the comfort

and reassurance of obtaining full reliable coverage throughout its

site. This would ensure an excellent Return on Investment when

considering the following short and long term benefits:

⇒ Excellent system integrity

⇒ System architecture scalability

⇒ Extensive features and call-centre functionalities

⇒ Future-proof technology with reduced upgrades and no

licensing costs

Further scope can be found within the advantages of streamlining

an existing phone system into a single IP-based reliable and cost-

effective telephony solution, which would offer extensive scalability

and could be further expanded to unlimited coverage.

Unified Communications

O T H E R S E R V I C E S

Asterisk Consultancy

Asterisk System Design

Project Management

Procurement and deployment

Expert Advice

Strata information Technology Co.

[email protected]

www.sitec.ae

Asterisk or proprietary?

+966 2 6068755 Jeddah - KSA

1-)'/1)*()("-0&3*,()-30")'/603%3AB"

ASTERISK as a VOIP GATEWAY

Asterisk Start Service – it is service on deployment of VoIP Gateway at customer's site. VoIP Gateway provides connection of customer’s existing PBX infrastructure to VoIP service providers. Asterisk also allows employees to make secure calls from their mobile devices to an internal telephone network or can be used as a complete VoIP PBX.

Installation of Asterisk server as Voip gateway gives a number of advantages:

No need to allocate a separate physical server for implementation. No drivers or hardware compatibility problems. Virtual machines is easy to backup. Fast migration to another server.

Objectives

More efficient usage of customer’s telephone system. Set up of connection to VoIP service providers. Remote connection to corporate telephone network. Training in-house IT staff introduction on Asterisk and its virtualization.

Expected Results

Virtual server with CentOS 5.6 and Asterisk 1.8.8.0. H.323 or SIP trunk to the customer’s PBX. Publishing Asterisk to Internet. Configuring SIP trunk to VoIP provider. Configuring routing of incoming and outgoing calls. Set up user accounts with the possibility to establish connection from various devices. Configuring backup of virtual machine. Configuration Asterisk in-house and total remotely technical support.

Hardware Requirements

Customer infrastructure requirements:

Virtual machine based on MS Hyper-V or VMware ESX: CPU 2.0 GHz, 2 GB RAM 40GB disk.

Direct IP address on the second network interface of VM or ability to publish VM via MS TMG, ISA or other firewall.

A telephone station with H.323 or SIP support (MS OCS/Lync Server, Avaya ACM, Definity, IPO, CISCO CM / CM Express), and also possibility to use Asterisk as a standalone PBX.

Administrator account for customer's telephone station.

Software Requirements

Management Console MS Hyper-V or VMware ESX. Client device with OS Win, Android 2.1 and above or iOS.

PSTN

MS Lynks/OCSServer

Media Gateway

SIP Signaling and

RTP Media

ISDN PRI (E1/T1), ISDN BRI, Analog FXO/FXS

Big in features, small in price!IP/PBX VoIP packaged solutions

Customers shopping for PBX systems may be horrified to see the price tags for a small telephone system with voicemail. Now there is an alternative, an inexpensive, full-featured PBX

phone system that runs on a custom server sitting in your office. Set up can be done through any Web browser in a

matter of minutes and no phone training is required.

Some highlights: Unlimited extensions, unlimited voicemail Powerful AutoAttendant with Scheduler Telcommuters keep their extension Multiple offices ready Easy to manage via Web browser Compatible with analog and IP phones Works with regular phone network or VoIP

Corporate Healthcare Education Real Estate IP/PBX solutions forany industry

or profession!Engineering Legal Manufacturing

Remote Office

PSTN VoIP

Main Office

ACD

Voice MailWireless

Home Office

Telephone ServiceProvider

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.

KEY FEATURES

Auto AttendantUnlimited Extensions

VoicemailVoicemail to Email

ACDOperator Panel

MP3 Music-on-HoldScheduler

Paging/IntercomName Directory

Call ForwardDID

Web Control PanelDo Not Disturb

E911Call Monitoring

Call ParkingCall Queuing

Call WaitingDistinctive Ring

Call Detail Records Interactive Directory Listing

Call Forward on No Answer Call Forward Variable

AVAILABLE PHONES

GXP2000multi line

9133imulti line

P2000Wwireless

GXV3000videophone

Flash Operator Panel is a real-time web interface to Asterisk. You can see what all of your extensions, trunks, and conferences are doing.

Asterisk is fully capable of functioning as an iPBX. The only requirements are: an Asterisk unit (PC);LAN (Local Area Network)

and IP telephone sets or IP gateways for connecting analog phones.

Asterisk can work with several IP telephony protocols, such asSIP, MGCP, H323, SCCP (Cisco’s proprietary protocol).

Since it works with analog and digital telephony protocols as well as several IP protocols, Asterisk can also be used

as a gateway between different protocols.

Streaming Media Access Supervised Transfer Talk Detection Text-to-Speech Three-way Calling Time and Date Transcoding Trunking VoIP Gateways Overhead Paging Protocol Conversion Remote Call Pickup Remote Office Support Roaming Extensions Route by Caller IDCall Recording Call Retrieval Call Routing (DID & ANI) Call TransferIVR Call Forward on BusyVoicemail Groups Web Voicemail InterfacePredictive Dialer

Copyright © 2006 RF Communications Inc.

Asterisk and the Asterisk logo are trademarks of Digium Inc.

Asterisk™ 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.

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™) Do Not Disturb E911 ENUM Fax Transmit and Receive 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 Integration of physically separate installations Uses commonly deployed data connections Allows a unified dialplan across multiple offices

Speech Cepstral TTS Lumenvox ASR Vestec ASR

Codecs ADPCM G.711 (A-Law & μ-Law) G.719 (pass through) G.722 G.722.1 licensed from Polycom® G.722.1 Annex C licensed from Polycom® G.723.1 (pass through) G.726 G.729a GSM iLBC Linear LPC-10 Speex

VoIP Protocols Google Talk H.323 IAX™ (Inter-Asterisk eXchange) Jingle/XMPP MGCP (Media Gateway Control Protocol SCCP (Cisco® Skinny®) SIP (Session Initiation Protocol) Skype UNIStim

Traditional Telephony Protocols 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. However, a patch is available)

ISDN Protocols AT&T 4ESS EuroISDN PRI and BRI Lucent 5ESS National ISDN 1 National ISDN 2 NFAS Nortel DMS100 Q.SIG