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