Asterisk Phone Systems
Download as PPTX, PDF1 like1,616 views
Asterisk is an open-source telephony software that started in 1999, transforming computers into communication servers and supporting over 500,000 downloads. It enables the creation of various telephony applications, such as PBXs, VoIP gateways, and call centers, making it a flexible and cost-effective solution for businesses of all sizes. Companies worldwide rely on Asterisk for its extensive features, including unified voicemail, conference bridging, and integrate with existing technology, all supported by a strong community and Digium's maintenance.
Asterisk Phone Systems
- 1.
- 2. Asterisk has wonlots of awards!
- 3. 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.
- 4. More information please… WhenPBXs 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.
- 5. The Case forAsterisk: 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:
- 6. PTSN ` Single Phone, SingleUser (Answering Machine) PTSN ` Three line, Five User (SOHO) Hanover Lebanon Chicago Enterprise User
- 7. But why doI 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
- 8. What can Asteriskdo 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.”
- 9. What type ofBusiness 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.
- 10. Sounds good, sowhat 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
- 11. What can TWdo 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)
- 12. What other TWProducts 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
- 13. 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)
- 14. More words ofRoy 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)
- 15.