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Information Technology

ASTERISK Open Source Telephony

Asterisk is the worlds most popular and powerful open source telephony p l a t f o rm .

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Lorem ipsum dolorsit amet, conse ctetuer adipiscing elit, sed diam nonummy nibh With Asterisk virtually any computer can become a telephony application server, drastically Feugait nulla facilisi. Lorem ipsum dolorsit amet, conse ctetuer laoreet dolore magna aliquam erat volutpat. euismod elit adispe.

reducing the cost of ownership and delivering a noticeable return on investment that no proprietary solution can match.

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Open source Asterisk works for SMBs, Corporate Enterprises,

Government and Contact Centers, offering robustness, advanced features and absolute flexibility and scalability.

Asterisk is the product of ten years of work by a community of thousands from around the globe. The community is made up of users, developers and advocates who have contributed their time and efforts to

Asterisk Consulting provides solutions to improve organisational efficiency by implementing managed 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 superb functionalities, with no licensing costs. We have a full team of in-house accredited dCAP (Digium Certified Asterisk Professional) and SIP Master engineers!

Get the most from your Asterisk solution

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 proprietary systems. The software is free and runs on inexpensive Linux servers, which also makes it the cheapest telephone system on the market today!

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Unified Communications
Asterisk offers an Open Source, cross-platform Instant Messaging 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

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

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

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

BENEFITS

Asterisk or proprietary?
The main scope of any organisation choosing to migrate from their

OTHER SERVICES
Asterisk Consultancy Asterisk System Design Project Management Procurement and deployment Expert Advice

No Licensing costs Significant call savings Improved SLA levels Enhanced productivity Open Source platform Supports the use of existing servers to comply with EU recycling guidelines

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 costeffective telephony solution, which would offer extensive scalability and could be further expanded to unlimited coverage.

Join the Asterisk Open S o u rc e C o m m u n i t y

Strata information Technology Co.


+966 2 6068755 Jeddah - KSA

sales@sitec.ae www.sitec.ae

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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 customers 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 customers 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 customers 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

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

SIP Signaling and RTP Media MS Lynks/OCS Server Media Gateway

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

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

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

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)

Speech
Cepstral TTS Lumenvox ASR Vestec ASR

Codecs
ADPCM G.711 (A-Law & -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

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

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