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Telephone Wreckers tells you all about Asterisk phone systems - the benefits, features and what product you'll need to build your own custom IP phone system.
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PRODUCT AND BRAND PRESENTATION
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Asterisk has won lots of awards!
What is Asterisk?
The Asterisk project started in 1999 when Mark Spencer released the initial code under the GPL open source license. Since that time, it has been enhanced and tested by a global community of thousands. Today, Asterisk is maintained by the combined efforts of Digium and the Asterisk community.
Asterisk is a Computer Based Telephone System software. At it’s core, Asterisk is an open source software that transforms an ordinary computer into a communications server. Since Spencer released Asterisk to the world in 1999 as a phone operating system, it has been downloaded 500,000 times, and it continues to be downloaded 1,000 times per day.
Digium is the inventor, sponsor, and maintainer of Asterisk open source and UC phone systems for SMBs.
More information please…
When PBXs were originally developed, wireline phone calls were the only type of electronic communication available.
Today, the communications landscape has expanded to include email, instant messaging, video conferencing, desktop sharing, SMS and mobile telephony.
Unified Communications is a catch-all term that describes the process of merging all of these technologies and integrating them with business processes.
Unified Communications aims to increase efficiency while simplifying management.
The Case for Asterisk:You can use Asterisk to build communications applications, things like business phone systems (also known as PBXs), call distributors, VoIP gateways and conference. Asterisk includes both low and high-level components that significantly simplify the process of building these complex applications.
Currently boasting over two million users, Asterisk supports a wide range of TDM protocols for the handling and transmission of voice over traditional telephony interfaces, featuring VoIP packet protocols such as SIP and IAX among others. It supports U.S. and European standard signalling types used in business phone systems, allowing it to bridge between next-generation voice-data integrated networks and existing infrastructure
Thousands of companies in more that 170 countries rely on Digium phone systems, here are but few:
0
PTSN
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Single Phone, Single User (Answering Machine)
PTSN
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Three line, Five User (SOHO)
Hanover
Lebanon
Chicago
Enterprise User
But why do I want Asterisk?
Cheap!
Flexibility
End user can control phone system
Highly extensible
Broad feature set
Powerful and efficient
Hardware is cheap, reliability and scalability.
Runs on Linux
What can Asterisk do for you?
Unified Voicemail
Advanced Meetme conferencing – conference bridging
Contact Centre Queuing
Interactive Voice Response
Automated Attendant
Video Conferencing
Jabber / Google Talk integration
Find me / Follow me
Out of state DID’s (all VoIP systems)
Call monitoring and recording
“Asterisk makes it simple to create and deploy a wide range of telephony applications and services, including IP PBXs, VoIP gateways, call centre ACDs and IVR systems.”
What type of Business can benefit from using Asterisk?
Commercial offices with between 2 and 2000+ extensions
Call centres, requiring call queuing (inbound and outbound) and call recording, IVR, ACD. When Asterisk was tested in 400-person call centers, costs fell by two-thirds.
Emergency Call centres (000)
Large sites requiring DECT solutions that roam across multiple cells
Hotel and billing solutions
The ability to block UNLIMITED numbers from being contactable.
Sounds good, so what do I need to know to use Asterisk?
Lets think of Asterisk like an engine / vehicle metaphor. Asterisk is an engine. It’s powerful. It’s flexible, it has
ENORMOUS potential.
What you require is a skilled engineer who can take the engine and build it into a vehicle.
A custom vehicle built specifically for your business needs and requirements.
Asterisk requires some fairly advanced technical skills, including a good working knowledge of IP networking, Linux system administration, traditional telephony experience and script programming know-how. Even experienced programmers will need to overcome learning curves to create a working system.
Ports also available for OSX and Windows
What can TW do for Asterisk?
AASTRA is a Participating Company of Asterisk
Starting with the Enterprise which is probably the system most Asterisk adaptable.
2-300 SIP extentions
60 concurrent calls
No limit SIP trunks
Auto-attendant
Voicemail to e-mail
500GB / 4 million minutes of auto recording (that's more than 7 and a half years of non-stop talk!)
Unified messaging
8 port FXO/FXS/BRI expandability
No licence fees
BRI available
Data rack mountable (brackets included)
What other TW Products and Asterisk work together?
IP VoIP Systems: Aastra / NEC / Aristel IP handsets new and refurbished:
Cisco IP Phones (except IP 7920)
Polycom IP Phones
Snom IP Phones
Avaya IP phones: DIGIUM HANDSETS WILL SUPERCEDE Headsets – Specific for Unified Communications
Blackwire C710 / C710-M
Blackwire C720 / C720-M
ENTREA C610 / C620
SAVI W0430 OTE Dect Conference Phones (Polycom / Plantronics Calisto) Call Recording Devices Music on Hold players UPS
Words of Roy 3G fail-over can be implemented if required for a third redundancy connection
VOIP Carrier/Provider can redirect VOIP Calls/Services to Mobile Phones in the event of ALL of the above failing for an extended period of time
Un-Paralleled Call Quality/Clarity
Video Calling (H263 and H264 supported) for Video Handsets and SIP Softphone Software such as CounterPath Bria and Polycom RealPresence
YOUR own custom company logo on most VoIP handsets
Scalable to meet expanding needs - Limited only by server performance and internet bandwidth
License-Free Voice-mail for all staff/extensions - even accessible remotely from external phones or from Skype
License-Free Extensions - virtually unlimited extensions
License-Free Conferencing - with pin security if required
Free Intra-ITSP Calls (both to End Points and other organisations)
Distinctive Ring to differentiate between INTERNAL and EXTERNAL calls
License-Free ACD Queues/Groups (Automatic Call Distribution) - sales/support/help-desk queues etc
Call Parking/Hold Queues
Call Monitoring/Recording for both Call Quality/Monitoring (listen only) and also staff Training/Coaching (listen and 1-way chat)
Handset convergence - Access your calls anywhere on any device - Hot-Desk, VOIP Handset, Soft Phone on PC, Soft-phone on Android/iPhone/iPad, etc
Skype integration with SKYPE-SIP Services (supports G729 low-bandwidth codec) for incoming Skype users to call your PABX directly from Skype software (Skype-based National/International VoIP Roaming for Staff and Clients)
Skype Calls TO YOU can be initiated directly from your Web Site via a Web Browser (for callers that have skype installed)
End User - Remote voicemail retrieval - via Dial-in, via Skype Dial-in, or email attachment
End User - Web-Based real-time portal/console/panel (FOP2 Virtual Switchboard/Attendant) for (phones/calls/lines/users status) and user functions (supports Terminal Server, Touch Screens AND Android WiFi Tablets)
More words of Roy
End User - Web-Based user-to-user IM/Chat within the User FOP2 Virtual Switchboard/Attendant (supports Terminal Server, Touch Screens AND Android WiFi Tablets)
End User - Polycom Desktop Connection for VVX SIP Video Handsets (PDC) Software (Workstations only - NOT RDP/Terminal servers) for extending your Polycom handset to your Computer screen
Visual/Interactive(mouse-click dialing) Directory via Web portal/console/panel (FOP2 Virtual Switchboard/Attendant) for both Company-Wide *and* Private contacts
Speed-Dial directory available for Company-Wide dialing (where hard-copy printed wall-lists are preferred) Incoming Caller ID Number to NAME IDENTIFICATION for incoming calls (based on above directory lists) Dictation - Dictate via PABX headset/handset to email as a sound file for processing by PA/Typist or Voice-to-text software Admin User - Web-Based real-time portal/console/panel (FOP2 Virtual Switchboard/Attendant) for company wide visibility
and admin functions (supports Terminal Server, Touch Screens AND Android WiFi Tablets) Direct In-Dial for incoming calls to bypass reception if required Auto Directory Attendant - IVR Voice-based directory of all staff names if required - even for external callers (this prompts
you to key in letters of names from phone keypad) Low Bandwidth Codecs (including G729) - optional - These should ONLY be used when ABSOLUTELY necessary to avoid
unnecessary server trans-coding/cpu load Fax Support (T.38 only - ITSP-Specific) - requires EITHER a T.38 gateway device OR T.38 FaxVoip Software as a DIRECT
SIP/T.38 CONNECTION to ITSP (SIP/Internet Telephony Services Provider) Manual override for automatic after hours/holiday queue processing Intercom feature - extension-based paging with auto-answer Call Logging and Reporting Least-Cost call routing for multiple carriers with mixed rates Flash Operator Panel (for ALL Network Users) example Receptionist Extension Status Monitoring (using sidecar multiple button expanders) Receptionist Enhanced BLF (Busy Lamp Field) for remote pickup of monitored ringing extensions (Additional LICENSED
OPTION)
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