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FreePBX is a full-featured PBX web application. If you've looked into Asterisk, you know that it doesn't come with any "built in" programming. You can't plug a phone into it and make it work without editing configuration files, writing dialplans, and various messing about.
FreePBX simplifies this by giving you pre-programmed functionality accessible by a user-friendly web interfaces that allows you to have a fully functional PBX pretty much straight away with no programming required. Some of the features that FreePBX supports out of the box are:
- Unlimited number of Voicemail boxes
- "Follow Me" functionality
- Ring Groups with calls confirmation (so if, eg, a cellphone is out of range and diverts to voicemail, all the other phones keep ringing)
- Unlimited number of Conferences (limited by available CPU power - about 300 simultaneous users in conferences on a P4 3ghz - 600 with a dual core!)
- Paging and Intercom functionality for man SIP phones that support it.
- Music on Hold (via MP3s, or streamed off the internet)
- Call Queues
- And many other features
FreePBX is built on the LAMPA™ stack (Linux, Apache, MySQL, PHP and Asterisk). It's a modular system, with click-to-install plugins downloadable over the internet from the online module repository.
FreePBX Features at a Glance:
- Add or change extension and voicemail accounts in seconds
- Native support of SIP, IAX, and ZAP clients (other endpoints are supported through custom extensions)
- Supports all Asterisk supported trunk technologies
- Reduce long distance costs with LCR
- Route incoming calls based on time-of-day
- Create interactive Digital Receptionist (IVR) menus
- Design sophisticated call groups
- Manage callers with Queues
- Upload custom on-hold music (MOH)
- Search company directory, based on first or last name
- Detect and receive incoming faxes
- Share administrative duties
- Backup and Restore your system
- Save audio recordings of calls
- View call detail reporting with asterisk-stat
- View extension and trunk status with Flash Operator Panel
- View conversation recordings with Asterisk Recording Interface (ARI)
Project Sponsored by SANGOMA
Service ProvidersFreePBX Services - AristoCraters offer a wide range of services for FreePBX Solution, including, setup and installation, configuration, customization, upgradation, troubleshooting, performance enhancement, and ongoing support to keep your FreePBX solution up and running.
VoIP / SMS / API - Unlimited Channels with measured rates and always free to test.
http://www.asterisksiptrunking.us Asterisk SIP Trunking - US — Offers SIP Trunking for Asterisk. Over 500,000 DID's available in 9,500 rate centers. You can activate and setup service in minutes. TDM Enterprise quality. Fully qualified Asterisk consultants ready to assist you. Live customer service with 24/hr ticketing system. No per channel fees, we offer unlimited channels in with your SIP trunk. FREE API for your website. Use our API to leverage the power of our customer user portal on your own website. You can build your own back office admin panel with our API and also provide your customers the ability to order DIDs in REAL TIME along with setting up SIP Trunks. Automate everything and increase customer base.Our Asterisk SIP Trunks are not only for Asterisk. We work with every IP Ready telephone system. Try us out today! No contract and low rates. DIDs are $0.50 per month, Toll Free DIDs are $0.50 per month US/Canada calling rates per minute are $0.009 for TIER1 quality calls. Nothing will route out low quality networks. International calling is available immediately upon registration. Auto refill on your credit card will ensure your account never runs out of funds.
DIDForSale SIP Trunking | DIDForSale provides largest coverage over US, UK and Canada. Offer Unlimited SIP Trunking Service for businesses.
- VoIP Services in US, UK and Canada
- Local Phone numbers in 11000+ rate centers
- Flat rates SIP Trunking, no contracts
- Unlimited SIP Trunking
- CNAM and 911
- _Pay as you go__
- NO Contracts
- instant activation
- Pay only for minutes you use
- DIDForSale SIP trunking works with any SIP PBX, including open-source pbx like freepbx, freeswitch, asterisk, opensips etc..
- Project website: Sip trunks easily configured through a FreePBX module. Brought to you by the Sponsors of FreePBX
- FreePBX Wiki
- FreePBX Resellers
- Asterisk GUI
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