Showing posts with label Source. Show all posts
Showing posts with label Source. Show all posts

Monday, February 7, 2011

Asterisk SCF: Make Open Source Cloud Telephony Happen

It’s been two months since Asterisk SCF was announced in AstriCon 2010. It really brings a lot of expectation to the open source community and it’s worth our waiting. Asterisk SCF is still in heavy development and will not be expected to be beta-level software until some time in 2011. But are you ready when it hits the ground? If you would like to start developing on Asterisk SCF, please Visit the Asterisk SCF wiki  to find information on downloading and building the Asterisk SCF source, documentation on the architecture, and see the Roadmap.

asterisk-scfAsterisk SCF is a framework that allows developers to create real-time communications applications that include voice, video and text and that meet the demands of a full range of uses, from embedded applications to enterprise and carrier solutions. Asterisk SCF is architected to provide the highest levels of availability, scalability, extensibility, fault-tolerance and performance.

Asterisk SCF will be delivered as a system of distributed components that can be deployed in clusters on a single system or on many systems, transparently. The Asterisk SCF platform will support, as a part of its basic architecture, the full range of real-time IP communications, including video, multi-channel wideband and ultra-wideband audio, chat, desktop sharing and other media types that may arise in the future.

Asterisk SCF is not a replacement for Asterisk, the world’s most widely used open source voice communications platform. Digium and the Asterisk community are committed to the continued development and support of Asterisk, the telecommunications software.

asterisk · Asterisk SCF · AstriCon 2010 · digium · embedded application · open source

<< More Than Two Million Downloads of Asterisk in 2010

The First Open Source Telephony Conference in 2011 >>

Saturday, January 1, 2011

Asterisk Open Source Billing Systems


A2Billing, combined with Asterisk is a physical Telecom Platform and Soft-Switch providing a wide range of telecoms services using both traditional telephone technology or VoIP. It contains a real time billing engine which rates and bills and invoices calls, and supports payment gateways.

This now gives any Telecom company a very good reason to consider the A2Billing Platform over the traditional offerings for TDM and VoIP Soft-Switches as well as wholesale billing.


Solutions


A2Billing can be used in a number of different roles, and our consultancy services can advise on the appropriate hardware and configuration of your Switch.


  * Calling Card service with either PIN or Caller ID recognition
  * Call-back service
  * VoIP Billing
  * VoIP termination for IP PBX systems
  * Wholesale VoIP Termination and Origination
  * Residential VoIP Termination and Origination.
  * Special Applications Platforms
  * Predictive Dialler and Sales Campaign Tool
  * Hosted PBX, IP Centrex and Multi-tenant systems.    
  * VoIP reseller white label solutions.    
  * and much more :)


There is online demo available at http://demo.asterisk2billing.org/a2b/. For more information about any of the features and benfits of any of our products, please contact us at sales@star2billing.com, or go to www.a2billing.net, and www.star2billing.com

AstBill is not only a web-based, user friendly billing interface for Asterisk and VOIP. It is also a Asterisk configuration and GUI management tool and a standardized implementation of Asterisk using REALTIME and static configuration as you please. AstBill is Open Source and under constant development. Prepaid, Postpaid and Calling Cards supported.

Asterisell is a open source web application for
rating,
showing to customers,
and billing Asterisk VoIP calls.
EasyITSP an open source ITSP/Hosted PBX solution that sits on top of Digium's open source Asterisk PBX, has been released to the public. EasyITSP allows people with little experience to easily manage and run an ITSP and/or Hosted PBX services with its small footprint and its easy to use admin and customer portals. EasyITSP is quickly gaining ground in its market with constant development of new features. To learn more and view a demo, visit http://lemens-ts.com/easyitsp
Freeside is the open-source billing, ticketing and account administration package for Internet Service Providers that now includes support for VoIP CDR rating and billing. Freeside supports Asterisk and many other CDR formats.

Freeside Internet Services also offers installation, integration, configuration, migration, training, and customization services.


Flexible pricing and rating plans including anniversary billing, pro-rating, usage based billing and more. Plug-in system allows easy addition of new pricing plans. VoIP rating and billing, including DID inventory, rate plans, free minutes, caps, and 800# billing. Real-time credit card and electronic check processing with all major processing gateways. Email, fax, printed and online invoicing.
Complete, integrated trouble ticketing system included, based on Request Tracker from Best Practical Solutions Auto-responds to customer email, assigns a ticket number and allows your staff to track requests collaboratively Multiple queues, custom fields, templates, ticket associations, templates, customizable workflow, and more.
Online signup page with immediate credit card authorization and provisioning. Customer self-care including invoice viewing, one-time and recurring payments, adding and removing services, and password changes. Simple API available to write your own self-service pages or applications.
Unlimited resellers with branded invoices, individual pricing and payment gateways. Virtualized reseller access allows resellers access to only their own customers. Flexible, fine-grained access control
Features:
Provide call rating for Asterisk. Calling Cards Integrates with AMP (amp.coalescentsystems.ca) An "unlimited" amount of different price lists. Batch billing Realtime billing Rates are billed to 6 decimal places. Rates are highly configurable. Import rate lists in cvs format. Variable billing increments. Call setup charges. Resellers. Least Cost Routing Web interface provides: - Card balances. - Ability to configure routes. - Ability to configure most of ASTPP. - Setup Brands. - Many other features. Interface for customers to view their calls. GPLed Source Code Intregrated with OSCommerce!! CitrusDB is an open source customer care and billing solution that can be used in 
many different service industries, including internet service, consulting, webhosting, telecommunications, and voip.
customer details unlimited services create invoices credit cards batches support notes under GPL v2 open source license Asterisk Hotel Billing is a system to bill the calls of rooms in hotels. It's very easy to use. 
Supports more than 15000 phone destinations. You can create PDF with detailed list of the calls, at the
moment the customer leaves de room, then the system starts a new bill for the next customer.
The application is in Spanish.
Requirements:
Asterisk + SQL CDR (Mysql tested) Web server with PHP + PEAR DB SIP/IAX Room extensions (not Zap) Zap/SIP/IAX trunk lines Features of the Asterisk hotel management system include:
   * Rooms inventory management with bulk FreePBX extension import
   * Call management by prefix with per minute and fixed rates
   * Check-In and Check-Out with associated actions
   * Billing with search by dates and/or by rooms with call detail
   * Integrated with Hotel wake-up module with web interface
   * Removal of voicemails and wake-ups on room check-out

Requirements:

Tool to automatically assign calls to users of a shared line and spread their phone bill
works currently for German customers of the Deutsche Telekom (using "Rechnung Online"), other companies that provide online bills could easily be added automated billing and accounting comparison of alternative billing plans PHP-interface Tool to access and rate in real-time Call Details Records generated by SIP Express Router, MediaProxy, Asterisk and other RADIUS gateways. - by Adrian Georgescu. CDRTool provides real-time mediation including normalization and rating for Call Detail Records generated by SIP Proxies and VoIP gateways. Combined PSTN rating based on time/destination with NGN rating based on traffic/application provides the ideal billing platform.
astCDRview is an open source, licensed under GNU/GPL, lightweight, Web-based, multi-language Asterisk SQLite CDR viewer that supports multiple outgoing carriers, multiple incoming numbers, billing, an address book, and extensions
http://www.open-rate.com License: Dual Licensed GPL/Proprietary OpenRate is an open source, commercially supported, high performance batch and real time rating engine. For end users it is licensed under GNU/GPL, for systems integrators and resellers the proprietary license allows white-labelling or re-branding.
Can support virtually any rating scheme using a pipeline configuration of standard modules. Extendible and highly flexible.
SQLBill is a Radius billing engine for the GNU Gatekeeper (GnuGk). It supports Radius or can be used directly on top of Postresql. It is a reliably backend system, but lacks a pretty frontend.
VOIP Billing - A large list of providers including both open-source and commerical billing solutions Open Source VOIP Software - Open-source software for VoIP, not necessarily billing-related

Tuesday, December 21, 2010

The First Open Source Telephony Conference in 2011

The real beautiful thing about open source telephony is when everybody talks about it and benefits from it. I am happy to know that the VoIP88.com has announced the 1st open source telephony conference in Shenzhen China in the coming year. And I do think it’s also the 1st one in the worldwide community since it will be launched on 10th Jan, 2011.

voip88_logo

VoIP88.com is now the largest open source telephony technical forum in China and is authorized to release the Elastix Chinese Version to the community. They are now providing free technical support in its forum and Elastix solution nationwide in local language.

What to Expect in the Conference?
The first Open Source Telephony Conference will be focus on promoting the most popular open source telephony software in China including Asterisk, AsterCC, Elastix, freeSWITCH, FreePBX and so on. Meanwhile, you will also expect to meet the new solutions and products provided by the hardware manufacturers such as Digium, OpenVox, Sangoma.

Who to Meet in the Conference?
Well, you will meet VoIP providers, PBX developers, Call Centre Developers, System Integrators of Open Source Technology, Asterisk users, hardware providers.

Conference Speakers:
Doug Vilim from Sangoma Technologies Corp
Solo from AsterCC
Tony Zhang from VoIP88.com

Click HERE for the original conference news in Chinese.

AsterCC · asterisk · Asterisk China · China Asterisk Conference · digium · elastix · freepbx · freeswitch · open source · OpenVox · sangoma · telephony · VoIP · voip88

OpenVox Announces Free Support for Asterisk Configuration >>

Thursday, December 9, 2010

Gitex 2010:The Rise of Open Source Telephony in Middle East

You are expected to see a new face in Gitex 2010 Dubai this week: OpenVox has been presented in the Zabeel Hall in booth Z-P31. It’s the first time that OpenVox shows up in the Gitex in the Middle East Marketing.  And the some of OpenVox newly released products is being presented in this show including AE2410P, IPC100 for a better choice to build a best cost effective Open Source telephony solutions.


gitex-2010 GITEX TECHNOLOGY WEEK is at the heart of the global technology network. Experience a spectacular celebration of interactive features, a showcase of leading technology brands and an unrivalled range of networking opportunities as GITEX celebrates 30 years of industry integration and leadership. Review and select from thousands of leading technology brands and industry experts.


Who else is to see in Gitex 2010? DVCOM is presenting Digium, Snom,Patton in its booth. You are not limited with your choice of asterisk products since OpenVox has already been here for you.


Sunday, September 26, 2010

Veteran Open Source Technologist to Keynote AstriCon 2010

Mark Summer

Mark Summer explores this year’s conference theme, “Asterisk: The World is Calling,” by sharing his experiences using the telephony platform to help improve local economies and quality of life around the world

Asterisk is changing the world, not just in the limited confines of  business-oriented UC communications, but in real, tangible, and life changing ways.  Our featured AstriCon keynote is  Mark Summer from Inveneo,  whose work in bringing Open-Source to under-served areas is literally changing lives.

The ability to communicate via voice is something we take for granted, but in many areas of the world that basic technology has remained out of reach due to cost, complexity, and implementation difficulty with traditional telecommunications infrastructure.  The good news is that in the last several years, Open-Source software and hardware has made inroads into these previously unreachable areas, and we’re seeing how Asterisk is actually changing the quality of life in areas where it has enabled people to talk with each other at a distance.

Mark brings to this keynote address his real-world experiences with Open Source communications in developing areas, and has concrete examples of how free software and methods are not only creating economic opportunity, but uplifting whole communities with communication.  His work in Sub-Saharan Africa, Haiti, and elsewhere focus on the implementation of Open-Source telecommunications, and the changes he has witnessed when people are given the tools to exchange ideas and create opportunity.  Asterisk is a component in this change, bringing not only the ability to connect people together, but the ability to allow local administration and innovation in what was once a centrally-managed, monolithic, and slow-changing telecom world.

Register Now for AstriCon 2010

Thursday, September 16, 2010

iRedMail 0.6.1: Open Source Mail Server With Postfix, Dovecot, Amavisd, ClamAV, SpamAssassin, RoundCube On Ubuntu 10.04

Postfix

iRedMail is a shell script that lets you quickly deploy a Open Source Mail Server solution in less than 2 minutes.

It supports Red Hat(R) Enterprise Linux & CentOS 5.x, Debian 5.0.x, Ubuntu 8.04/9.04/9.10/10.04, FreeBSD 7.x/8.x, it supports both i386 and x86_64. Most components used in iRedMail are provided by Linux distributions officially, it means iRedMail users can get software update support for as long as for the distribution itself. It supports OpenLDAP and MySQL as backends to save the virtual domains and virtual users. It offers a web-based admin panel.

iRedMail was initially published in 2007, and now iRedMail 0.6.1 also supports Ubuntu 10.04.

Main components used in this solution: http://code.google.com/p/iredmail/wiki/Main_Components Discussion forum: http://www.iredmail.org/forum/ Project home page: http://code.google.com/p/iredmail/ Project DEMO: http://demo.iredmail.org/

 

1 Requirements

To install such a system you will need the following:

The Ubuntu 10.04 server CD, available here: http://releases.ubuntu.com/releases/10.04/ubuntu-10.04.1-server-i386.iso (i386) or http://releases.ubuntu.com/releases/10.04/ubuntu-10.04.1-server-amd64.iso (x86_64) It is strongly recommended to install a minimal Ubuntu system, and then deploy the mail service via the iRedMail solution.For 10.04 (Lucid, LTS), it is not needed to modify default /etc/apt/sources.list. iRedMail uses packages from the default stable version, both 'main' and 'universe' repositories .

 

2 Preliminary NoteIn this tutorial we use: Hostname: mail.example.com IP address: 192.168.1.10 LDAP suffix: dc

Wednesday, September 15, 2010

Meet The 2600hz Project, The New Sound of Open Source Telephony

I really miss my Bluebox and the days then that would work.  Gotta love the $5 rRdbox from hallmark or the infamous Blotto Box that was no joke at all.  We salute you Captain Crunch!
 
Gigaom - Some of the core developers behind FreePBX — a well-known, open-source phone system — have teamed up and started The 2600hz Project, a commercial entity promoting a collection of open-source telephony applications and libraries. Today, they are releasing blue.box, a reworked version of open source FreePBX. The new venture is co-founded by Darren Schreiber and is also a subsidiary of newly formed VoIP Inc. The 2600hz Project received $250,000 in funding from an unnamed investor.

2600 Hz is the frequency that the phone companies used back in the day and was hacked by those seeking to make free long distance phone calls. In order to do so, one needed a device that generated the 2600 Hz tone, called the blue box. The new venture is an homage to that heritage.

From what I understand, the new company was formed after some disagreements between the FreePBX developers and the original backers of the project. FreePBX is a graphical user interface that sits on top of open-source telephony software such as Asterisk. FreePBX was promoted by Bandwidth.com.

Click Here to Continue Reading

Saturday, August 28, 2010

Meet The 2600hz Project, The New Sound of Open Source Telephony

I really miss my Bluebox and the days then that would work.  Gotta love the $5 rRdbox from hallmark or the infamous Blotto Box that was no joke at all.  We salute you Captain Crunch!
 
Gigaom - Some of the core developers behind FreePBX — a well-known, open-source phone system — have teamed up and started The 2600hz Project, a commercial entity promoting a collection of open-source telephony applications and libraries. Today, they are releasing blue.box, a reworked version of open source FreePBX. The new venture is co-founded by Darren Schreiber and is also a subsidiary of newly formed VoIP Inc. The 2600hz Project received $250,000 in funding from an unnamed investor.

2600 Hz is the frequency that the phone companies used back in the day and was hacked by those seeking to make free long distance phone calls. In order to do so, one needed a device that generated the 2600 Hz tone, called the blue box. The new venture is an homage to that heritage.

From what I understand, the new company was formed after some disagreements between the FreePBX developers and the original backers of the project. FreePBX is a graphical user interface that sits on top of open-source telephony software such as Asterisk. FreePBX was promoted by Bandwidth.com.

Click Here to Continue Reading