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Wide Open VoIP: Top 50 Open Source VoIP Apps

Source: www.virtualhosting.com

For many businesses, open source VoIP programs and apps offer a great way to save thousands of dollars every year in telephony costs. Better yet, open source programs are fully customizable to a business’ specific needs, making them a popular solution that often just can’t be beat. This popularity hasn’t just helped business, but has also driven many open source programs to the forefront of their industry. In fact, it has been speculated that open source VoIP solutions could surpass the popularity of the ubiquitous desktop solution Linux. Here are a few of the open source programs and developers out there that have had loads of success as VoIP and open source solutions for it become more and more common in businesses around the world.

SIP Proxies

SIP Proxies enable service providers to build scalable and reliable VoIP networks that are based on the Session Initiation Protocol. This allows a full array of call routing capabilities that make the most of network performance. Here are some of the most popular and successful SIP proxies on the market.

  1. OpenSer: OpenSER has been described as a “mature and flexible” SIP server so it’s no surprise that it’s popular among users. OpenSER development began with FhG FOKUS SIP Express Router, but then diverged into its own feature-laden software package that was released in 2005. Since then it’s been exhibited around the world, and makes a great addition to Linux systems looking to employ VoIP technology.
  2. VOCAL: Open source VoIP developers can benefit from the software and tools found in VOCAL. Developed through the Cisco sponsored labs at Vovida, VOCAL is fully customizable to business needs and can provide call routing, billing information, call control and more in an easy to control and maintain Linux based system. It’s been successful due largely in part to its immense capability for adaptation and scalability, and likely will only see further integration into business systems in the future.
  3. partySIP: Developed back when VoIP was just starting to take off, partySIP can still be a relevant solution for businesses looking for VoIP today. This lies largely in the modular construction of partySIP, which relies on various plugins to add or remove capabilities. This flexibility allows its users to disable useless functions and enable new ones with very little development, making it easy to use and customize, which is likely the reason for users’ continued interest in the product.
  4. SIP Express Router: This high performance SIP product can act as registrar, proxy or redirect server depending on your needs. It’s been widely successful in the VoIP market due to its ability to deal easily with operational problems like broken network components. Another reason it’s loved is its scalability from small office environments to acting as a PBX replacement and can in many cases act as a replacement for the very popular Asterisk system.
  5. MjServer: One of the things that makes MjServer so important to the VoIP market is that it works on a variety of platforms, not just Linux, so those who aren’t quite ready to take the fully fledged open source route can ease into it. MjServer is a Java based application that is easily configurable and can act as a registrar, redirect or proxy in your VoIP setup, making it a versatile and useful tool for implementation.
  6. OpenSBC: OpenSBC has been in use for over 7 years in both low and high volume applications. In this way, it’s a very reliable system, but also still employs a great deal of possibility for expansion and modification based on personal needs for the program. In fact, like most open source VoIP applications, the developers actively encourage the changing and development of the program to make it better for all users.
  7. sipX: Developed by SIPFoundry, sipX is designed to be an incredibly feature rich and standards compliant infrastructure for businesses who want to employ VoIP technology. It is, in fact, one of the most widely used and well respected open source developments out there and feature wise is very similar to Asterisk.

SIP Clients

Session Initiation Protocol is a signaling protocol for Internet conferencing, telephony, presence, events notification and instant messaging, and is fast becoming one of the more popular protocols for VoIP in businesses and homes alike. Here are a few programs that have helped bring SIP to the forefront of the market.

  1. Linphone: Linphone is promoted as a solution to help users communicate more freely over the Internet using voice, video and text messaging. Recent updates to the program have made it even better, solving many compilation issues while adding improved interoperability and new features. While currently only stable on Linux systems, development is under way for a Windows version as well.
  2. PhoneGaim: If you haven’t heard of PhoneGaim you’ve likely heard of its proprietary counterpart Gizmo Project. While it doesn’t have the instant name recognition of its VoIP cousin Gizmo, PhoneGaim is still a product to take note of. Developed in an attempt to challenge Skype, the program is loaded with integrated features that help make the VoIP experience rewarding, even for those just using the software at home.
  3. OpenWengo: Started and developed by the French company, Wengo, OpenWengo is a great, and popular, open source choice for anyone looking for simple and easy-to-use VoIP software. This softphone program allows users to call between computers and phones, and has additional instant messaging and contact management capabilities. The recent development of a Firefox plugin that allows users to make calls quickly and simply from their browsers is just one example of the continued innovation and popularity of this multi-featured program.
  4. Cockatoo: Users of Thunderbird have Cockatoo to thank for simple VoIP integration with their email. The program allows users to make a call simply by clicking on entries in their address book. It’s simplicity and aim to make VoIP more fully integrated into business systems has made it a popular addition to business and personal computers.
  5. Minisip: Minsip is an Internet based phone that can be used to make phone calls, instant message and video call to anyone connected to the same SIP network. Developed by PhD and masters students at Royal Institute of Technology in Stockholm, Minisip is a simple by highly functional VoIP phone. Users can even make calls from PDAs or pocket PCs running Windows or Linux, making VoIP on the road easy and cheap.
  6. OpenZoep: Developed by Voipster, OpenZoep is a popular client-side VoIP choice, providing the ability to both make calls and send and receive instant messages. Since its release, developers have continually added new features, especially from users in Europe, where the product was first developed. Continued changes and a responsive market have made OpenZoep a popular solution both here and abroad.
  7. Shtoom: Shtoom is a open-source, cross-platform VoIP softphone, implemented in Python which also includes an application called doug which can be used to write and modify VoIP applications. This built-in framework for modification encourages customization, one of the reasons open source software is so popular.
  8. Twinkle: Linux users have embraced the softphone Twinkle for making VoIP calls through an SIP protocol. Twinkle is a great solution for many users as it provides many, if not more, of the features found in regular telephony including custom ring tones, voice mail, conference calling, and multiple lines. These features, in addition to its open source usability, make Twinkle a popular choice among Linux users.
  9. YeaPhone: YeaPhone is unique among open source VoIP systems in that it hopes to take the computer monitor and keyboard completely out of the picture when making VoIP calls, opting instead to use the Yealink USB headset. This makes it more similar to many commercially available phone systems, and a popular choice among users searching for an open source alternative to those systems.

H.323 Clients

H.323 is the traditional protocol for most VoIP systems which has been continually refined with new elements to help improve voice and video quality. These popular VoIP clients make the most of what H.323 is capable of.

  1. YATE: The YATE system relies on its ability to adapt to the conditions in which it’s being used. A flexible routing engine allows communications to be made efficiently and cheaply, both often big concerns to businesses when choosing VoIP platforms. It’s easily combined and expanded with other services making it an incredible versatile and successful tool in the VoIP market.
  2. FreeSWITCH: FreeSWITCH is “an open source telephony platform designed to facilitate the creation of voice and chat driven products scaling from a soft-phone up to a soft-switch.” This ability to work both as a soft phone and a soft switch depending on the needs of the users makes it an attractive and intriguing option for many searching for VoIP technology. It’s even been touted as a viable alternative to using Asterisk, as many plugins and modules are available that don’t require reworking the main program code.
  3. Eikga: Formerly known as GnomeMeeting, Ekiga is an open source VoIP and video conferencing program that was developed for the Linux GNOME platform. It has a relatively simple interface, but gains one major advantage with users in that it works with both the H.323 protocol as well as with SIP, giving it double the functionality for users.
  4. OpenPhone: The original goal of OpenPhone was to enable every computer on the planet with phone capabilities. While this lofty goal may still be a ways off from completion, the OpenPhone software is still making strides in that direction. With an active development community, OpenPhone is a great place to find open source inspiration and functionality.
  5. XMeeting: Mac users need not despair, there are great open source alternatives for you as well, like XMeeting. XMeeting is the first H.323 compatible video conferencing client for Mac OS X, and not only supports H.323 but SIP as well. This functional versatility has made it a great solution for businesses primarily using Macs but also wanting to make the most of VoIP technology.

IAX Clients

IAX stands for inter-Asterisk exchange and programs using this protocol are used to enable VoIP connections between servers as well as to facilitate client-server communication. Here are a few of the most popular of these applications.

  1. IAXComm: IAXComm is a multi-platform softphone that works with Asterisk to allow users to place and receive VoIP calls. With features like music on hold and speakerphone, it is a great addition to an Asterisk system for businesses looking for VoIP technology.
  2. Kiax: Kiax relies on the IAX protocol to help it make it friendlier to users behind a NAT, or a router system that rewrites the source and/or destination IP addresses of IP packets as they pass through. Kiax maintains a simple interface that allows users to make calls to an Asterisk server quickly and easily, helping speed the spread of VoIP implementation both in homes and businesses.
  3. YakaPhone: YakaPhone is a simple and skinnable softphone. It is often a great solution for businesses looking for VoIP technology that is relatively simple but also easy to customize and use for day to day business. Businesses can even customize the phone skins to reflect company logos and branding, making it a more personalized experience.
  4. SFLPhone: For those with busy or especially large offices, SFLPhone is often one of the best IAX options as it was designed to handle high loads of daily phone calls. New partnerships should take it even further into the enterprise sector, as it has been announced that video conferencing is in the works.

PBX and IVR Platforms

PBX, or private branch exchange refers to a the telephone operating systems of a business or office, rather than those established for public use. Part of those systems might include Interactive Voice Response, which allows the computer to detect voice and touch tones to route phone calls to the appropriate menus or locations. These VoIP programs have taken the lead in those technologies.

  1. Asterisk: Asterisk is perhaps the greatest open source VoIP success story of them all. It is the leading open source telephony engine and tool kit and is used in thousands of servers and VoIP setups all over the world. What makes it so great? The standard system supports many features available in proprietary PBX system like voice mail, conference calling, interactive voice response, and automatic call distribution but also has the versatility to be adapted and personalized based on business or individual needs.
  2. OpenPBX: Developed by Australian company Voicetronix, OpenPBX is a popular solution both with small offices and with large call centers. With features like unlimited voicemail, auto-attendant, automatic call distribution, music on hold and call parking it’s easy to see why. It also has the advantage of highly compact Perl code, meaning it’s very easy to customize and extend.
  3. GNU Bayonne: An integral part of GNU telephony, Bayonne offers users technology that is not only free but scalable and customizable as well. Working with the complete suite of GNU enterprise solutions, Bayonne can be an easy way for users to integrate with telephony and provides a great VoIP solution for many.
  4. CT Server: CT Server is based on the ccscript language developed by David Sugar for the IVR server Bayonne as well as Perl for other tasks like database lookup. CT Server has been great resource for developers looking for framework for customizing or creating their own PBX quickly and creatively.
  5. sipX PBX: One of the main competitors to Asterisk, the sipX PBX and Asterisk are often compared. In contrast with Asterisk’s complete open source approach, sipX has a bit more of a commercial flair, as additional support and plugins can be purchased from the developers website. But sipX, once installed on your system, can provide much of the same functionality and in some cases might even be easier to use.
  6. Trixbox: Fast becoming one of the most popular Asterisk based PBX phone systems, Trixbox has seen loads of success in the past few years from businesses and enterprises searching for a VoIP solution. Designed for businesses with anywhere from 2 to 500 employees, the product comes in a few different versions.
  7. Evolution PBX: Evolution is another, more commercial application based on the open source server Asterisk. Basic editions of the software are free, however, and can be downloaded from the developers site. Evolution has been instrumental in helping solve one of the major obstacles to many businesses implementing VoIP as it makes integrating existing phone systems with newer VoIP systems easier, making the change much more cost effective for businesses, a key selling point for any VoIP product.
  8. CallWeaver: Originally derived from Asterisk, CallWeaver works on many different platforms and with new versions being released regularly it has a growing list of features. CallWeaver was developed as an alternative form of Asterisk that encourages community involvement and employs multiple vendors who drive the project rather than just one working for a single interest. This open-minded approach to open source VoIP has won the program many fans who believe that it’s already better than other versions of Asterisk.

Stacks and Libraries

Stacks and libraries are an integral part of what makes open source such powerful technology. Using these resources, businesses or individuals can develop and refine VoIP systems that work best for their business. These are just a few of these such resources that have had a big impact on VoIP development.

  1. OpenSIPStack: OpenSIPStack provides developers with a platform agnostic stack implementation of RFC 3261 so that development can be done in Linux, Solaris, BSD, Darwin and Windows. This versatility has made it an ideal choice for developers wanting to work in a variety of platforms.
  2. The GNU oSIP Library: Developers wanting to work with SIP have found just about everything they need in this library. Described as having the aim to “provide multimedia and telecom software developers an easy and powerful interface to initiate and control SIP based sessions in their applications” the GNU oSIP Library can do just that as it includes not only a library but examples of programs that use the oSIP protocol to operate.
  3. Twisted: Twisted comes from Twisted Matrix Laboratories and is an “event driven networking engine written in Python.” It supports a variety of protocols ((including HTTP, NNTP, IMAP, SSH, IRC, and FTP) and also has support for SIP, making it ideal for VoIP development.
  4. Verona: The Verona Project is an open source VoIP toolkit based on a phone API called Phapi and a minimal user agent called aptly miniua. It is similar to the toolset used in the highly successful OpenWengo software but is modified to reduce dependence on certain libraries, allowing users reliable building blocks for constructing their own VoIP programs.
  5. PJSIP: Written in C, PJSIP is an open source protocol stack for SIP. Due to its small footprint, high portability, customizability, and loads of other features its become a popular choice among SIP developers.
  6. eXosip: The eXosip library is a common choice among those who want to take the complexity of using the SIP protocol for multimedia session establishment down a notch. eXosip hides it, and makes implementing SIP easier whether you’re using it for VoIP or for something like multiplayer gaming.
  7. Vovida SIP: Vovida is a hugely popular place to get VoIP software both to use as is and like this protocol stack, to be used more commonly in further development of VoIP programs. This SIP stack is popular with Linux based developers wanting to embrace this protocol.
  8. reSIProcate: Part of SIPFoundry, reSIProcate works in a variety of operating systems including Unix, Windows, Mac OS X and more. The application is well suited and widely used in companies wishing to implement phones, softphones, gateways, proxies, or instant messaging.

Developers

While anyone is able to edit and create parts of open source software, the original programming has to come from somewhere. These are a few developers that have had great success in creating and releasing many of the most popular and widely used VoIP technologies in the open source field today.

  1. SIPFoundry: SIPFoundry is a not for profit open source community that aims to support the development and adoption of the SIP protocol. It’s also the home of much of the development of the sipX PBX for Linux, an award winning open source PBX program. The success of the sipX project as well as the increasing popularity of SIP have brought the SIPFoundry to the forefront of the VoIP community.
  2. Pingtel: Pingtel’s unique approach to the VoIP market may have a lot to do with their success. Using a system that runs using Linux and the sipX, Pingtel hopes to give business more control over how VoIP is built and used within their communications, something that proprietary software often can’t offer. The company also prides itself on providing reliable support and service for their products, making many business more willing to use them as there is less risk if something goes wrong.
  3. Vovida: Vovida is home to numerous SIP protocol stacks to help developers create and innovate new VoIP technologies and programs. Acquired in late 2000 by Cisco systems, this company’s work is well funded and its VOCAL tools and software have helped push VoIP development forward.
  4. Sangoma: Sangoma is a Canadian based company that develops both hardware and software based on the open source model, especially that having to do with telephony. While popular in North America, Sangoma is capitalizing on the hotbed of tech activity in Asia by forming a partnership with Vietnamese telephone distributor Dinh Quang. Their extension of open source VoIP software into new and widely used markets made them one of the most successful VoIP open source developers of 2007.
  5. Digium: With over a million downloads, Digium is one of the leading providers of Asterisk’s open source PBX software and has been the recipient of several awards for best open source software. With continued growth, and the acquisition of smaller VoIP players like Switchvox, Digium continues to add to its VoIP arsenal and likely will remain at the forefront of VoIP developers in years to come.

Miscellaneous

VoIP provides an opportunity for many different types of open source development to improve and refine systems. Here are a few miscellaneous programs that aren’t directly providing VoIP service, but are having an impact on the technologynonetheless.

  1. SIP Thor: SIP Thor is based on P2PSIP technology, and is built so that there is no single point of failure despite a large amount of scalability. With these features as well as quick disaster recovery and reliable service, those looking to start a VoIP reselling venture have found SIP Thor to be a great choice.
  2. MobiCents: MobiCents is billed as “the most popular Open Source Service Logic Execution Environment (JSLEE) and SIP Application Server for the Java platform.” MobiCents facilitates the creation of new services, enabling the development of a market oriented and cost effective platform, all the while encouraging developers to continue coming up with new and better ideas.
  3. Aradial: For business needing a means to bill minutes used with VoIP Aradial provides a viable open source solution. It’s easy to use servers are highly scalable and feature a plug-in architecture for quicker upgrades. Its low cost and easy adaptability make profit margins higher for businesses reselling VoIP and make it a popular solution.
  4. Lintad: Lintad is helping to make VoIP more than just a voice technology. The program provides both fax and voicemail support for VoIP phone systems. While voicemail is nothing new, the fax capabilities are nice addition and one that makes switching to VoIP much less painful for businesses.
Published on February 20th, 2008 under , , , , , ,

Zoiper Free for all major platforms, Linux, OS X and windows

Source: snapvoip.blogspot.com

Attractel has released two versions of Zoiper IAX and SIP softphone: Zoiper 2.0 for Linux and Zoiper 2.0 for Mac Os X. (There is also Zoiper 2.0 for Windows.). These are all free versions. In addition to free versions, there is also a BIZ version available as well. The BIZ version has more options and you can compare those here.

Following is from release notes;

Linux, ver. 2.0:

• RTP media address is set automatically
• Implemented per account options (STUN for SIP accounts and codecs)
• Fixed problem with the audio
• Fixed bug: when dial is clicked and no account is selected, Zoiper crashes
• Fixed bug: when deleting last account, pressing OK crashes the phone
• Fixed creating of new accounts- empty name is not permitted anymore

Published on October 10th, 2007 under , , , , ,

Something Old (H.323), Something New (IAX), Something Hollow at BlackHat

Source: snapvoip.blogspot.com

"Something Old (H.323), Something New (IAX), Something Hollow(Security), and Something Blue (VoIP Administrators)"Presenters: Himanshu Dwivedi and Zane LackeyThe presentation will discuss the security issues of two VoIPprotocols, including H.323 and IAX. The talk will discuss theauthentication and authorization weaknesses of each protocoland how it affects the overall VoIP network. Four new VoIPsecurity testing tools will be released during thispresentation, focusing on H.323 and IAX attacks.About iSEC Partners, Inc. (http://www.isecpartners.com)iSEC Partners to Present Six Talks at Black HatSAN FRANCISCO, July 18 /PRNewswire/

Original News Release by PRNewswire.
Published on July 18th, 2007 under , , , , ,

Asterisk: The Future Of Telephony under under the Creative Commons license

Source: snapvoip.blogspot.com


O’Reilly Media has released the Asterisk: The Future Of Telephony under under the Creative Commons license. Kudos goes to O’Reilly Media and the three authors, Jim Van Meggelen, Jared Smith, and Leif Madsen. Download links and Authors links are under the links at the bottom of the post.
Once I know the rules, I will post it in IPTELEPHONY in Google Groups.
Asterisk: FOT has received wide attention in the VoIP and IPPBX realm. It is a very well put together book that appeals to new comer to Asterisk as well as to the IPPBX pro. For information, I have listed the contents of the book below.

Contents of the Asterisk: The future of Telephony
Foreword

Preface

1. A Telephony Revolution
VoIP: Bridging the Gap Between Traditional Telephony and Network Telephony
Massive Change Requires Flexible Technology
Asterisk: The Hacker’s PBX
Asterisk: The Professional’s PBX
The Asterisk Community
The Business Case
This Book

2. Preparing a System for Asterisk
Server Hardware Selection
Environment
Telephony Hardware
Types of Phone
Linux Considerations
Conclusion

3. Installing Asterisk
What Packages Do I Need?
Obtaining the Source Code
Compiling Zaptel
Compiling libpri
Compiling Asterisk
Installing Additional Prompts
Updating Your Source Code
Common Compiling Issues
Loading Zaptel Modules
Loading libpri
Loading Asterisk
Directories Used by Asterisk
Conclusion

4. Initial Configuration of Asterisk
What Do I Really Need?
Working with Interface Configuration Files
FXO and FXS Channels
Configuring an FXO Channel
Configuring an FXS Channel
Configuring SIP
Configuring Inbound IAX Connections
Configuring Outbound IAX Connections
Debugging
Conclusion

5. Dialplan Basics
Dialplan Syntax
A Simple Dialplan
Adding Logic to the Dialplan
Conclusion

6. More Dialplan Concepts
Expressions and Variable Manipulation
Dialplan Functions
Conditional Branching
Voicemail
Macros
Using the Asterisk Database (AstDB)
Handy Asterisk Features
Conclusion

7. Understanding Telephony
Analog Telephony
Digital Telephony
The Digital Circuit-Switched Telephone Network
Packet-Switched Networks
Conclusion

8. Protocols for VoIP
The Need for VoIP Protocols
VoIP Protocols
Codecs
Quality of Service
Echo
Asterisk and VoIP
Conclusion

9. The Asterisk Gateway Interface (AGI)
Fundamentals of AGI Communication
Writing AGI Scripts in Perl
Creating AGI Scripts in PHP
Writing AGI Scripts in Python
Debugging in AGI
Conclusion

10. Asterisk for the Über-Geek
Festival
Call Detail Recording
Customizing System Prompts
Manager
Call Files
DUNDi
Conclusion

11. Asterisk: The Future of Telephony
The Problems with Traditional Telephony
Paradigm Shift
The Promise of Open Source Telephony
The Future of Asterisk

A. VoIP Channels

B. Application Reference

C. AGI Reference

D. Configuration Files

E. Asterisk Command-Line Interface Reference

Index

Links;
download book as a single entity,a PDF file.(4.5MB) USA1 USA2 UK NL
Download each chapter is a seperate PDF file (3.1MB) USA1 USA2 UK NL
O’Reilly Media
Jim Van Meggelen
Jared Smith
Leif Madsen

New GUI for Asterisk, Callware Voiceone.

Source: snapvoip.blogspot.com

A Source forge based Open Source project has developed an Management Gui for the IPPBX, Asterisk. I know all of you will say Go for Trixbox (I am already there ;)) but I want to give exposure to all applications that enhance Asterisk. I have not tried the product yet an all information I am providing is based on VoiceOne website.
The current release is 0.4.2. If you are a code Guru, you can grab the SVN version from the Sourceforge site.
From the outlook of the product, it seems promising. The interface is based on PHP and even though it is in it’s infancy many a features listed below are functional. They have an online demo that you could test out application and the function. Since these interfaces to Asterisk are personal tastes, some may prefer VoiceOne and asterisk to Trixbox!
Following are the feature list from the web site. So even if you already have Asterisk IPPBX management console, go check it out, give a boost to OSS developers.
* Client/Server architecture based on web services
* Relies on Asterisk real time Architecture (ARA) for database storage
* Two different panels, Personal for users and Configuration for administrators
* Extensions management
* Fully customizable users profile, including Voice mail, Call Forwarding ("Follow Me") and Do Not Disturb
* Highly configurable rule sets for outbound and inbound calls
* Static LCR (Least Cost Routing)
* Supports VoIP providers (SIP and IAX) and traditional Telco carriers
* Links remote offices via IAX with RSA public key encryption
* Powerful IVR creation system
* Queues management
* Conference rooms handling
* Sounds and Music On Hold management
* Applications and macros editor
* System Macros and Functions preloaded (DID/DDI, Call Back and DISA included)
* Plugins system to share ready-to-use macros and application with the VoiceOne community
* Powerful configuration of mISDN and Zap drivers based hardware
* Java SIP phone embedded
* I/O interface and PBX CLI (Command Line Interface)
* Static-like text editor for conf files

Links;
Callware VoiceOne
VoiceOne Demo
Quick Install Guide
Sourceforge Project

New GUI for Asterisk, Callware Voiceone.

Source: snapvoip.blogspot.com

A Source forge based Open Source project has developed an Management Gui for the IPPBX, Asterisk. I know all of you will say Go for Trixbox (I am already there ;)) but I want to give exposure to all applications that enhance Asterisk. I have not tried the product yet an all information I am providing is based on VoiceOne website.
The current release is 0.4.2. If you are a code Guru, you can grab the SVN version from the Sourceforge site.
From the outlook of the product, it seems promising. The interface is based on PHP and even though it is in it’s infancy many a features listed below are functional. They have an online demo that you could test out application and the function. Since these interfaces to Asterisk are personal tastes, some may prefer VoiceOne and asterisk to Trixbox!
Following are the feature list from the web site. So even if you already have Asterisk IPPBX management console, go check it out, give a boost to OSS developers.
* Client/Server architecture based on web services
* Relies on Asterisk real time Architecture (ARA) for database storage
* Two different panels, Personal for users and Configuration for administrators
* Extensions management
* Fully customizable users profile, including Voice mail, Call Forwarding ("Follow Me") and Do Not Disturb
* Highly configurable rule sets for outbound and inbound calls
* Static LCR (Least Cost Routing)
* Supports VoIP providers (SIP and IAX) and traditional Telco carriers
* Links remote offices via IAX with RSA public key encryption
* Powerful IVR creation system
* Queues management
* Conference rooms handling
* Sounds and Music On Hold management
* Applications and macros editor
* System Macros and Functions preloaded (DID/DDI, Call Back and DISA included)
* Plugins system to share ready-to-use macros and application with the VoiceOne community
* Powerful configuration of mISDN and Zap drivers based hardware
* Java SIP phone embedded
* I/O interface and PBX CLI (Command Line Interface)
* Static-like text editor for conf files

Links;
Callware VoiceOne
VoiceOne Demo
Quick Install Guide
Sourceforge Project

New GUI for Asterisk, Callware Voiceone.

Source: snapvoip.blogspot.com

A Source forge based Open Source project has developed an Management Gui for the IPPBX, Asterisk. I know all of you will say Go for Trixbox (I am already there ;)) but I want to give exposure to all applications that enhance Asterisk. I have not tried the product yet an all information I am providing is based on VoiceOne website.
The current release is 0.4.2. If you are a code Guru, you can grab the SVN version from the Sourceforge site.
From the outlook of the product, it seems promising. The interface is based on PHP and even though it is in it’s infancy many a features listed below are functional. They have an online demo that you could test out application and the function. Since these interfaces to Asterisk are personal tastes, some may prefer VoiceOne and asterisk to Trixbox!
Following are the feature list from the web site. So even if you already have Asterisk IPPBX management console, go check it out, give a boost to OSS developers.
* Client/Server architecture based on web services
* Relies on Asterisk real time Architecture (ARA) for database storage
* Two different panels, Personal for users and Configuration for administrators
* Extensions management
* Fully customizable users profile, including Voice mail, Call Forwarding ("Follow Me") and Do Not Disturb
* Highly configurable rule sets for outbound and inbound calls
* Static LCR (Least Cost Routing)
* Supports VoIP providers (SIP and IAX) and traditional Telco carriers
* Links remote offices via IAX with RSA public key encryption
* Powerful IVR creation system
* Queues management
* Conference rooms handling
* Sounds and Music On Hold management
* Applications and macros editor
* System Macros and Functions preloaded (DID/DDI, Call Back and DISA included)
* Plugins system to share ready-to-use macros and application with the VoiceOne community
* Powerful configuration of mISDN and Zap drivers based hardware
* Java SIP phone embedded
* I/O interface and PBX CLI (Command Line Interface)
* Static-like text editor for conf files

Links;
Callware VoiceOne
VoiceOne Demo
Quick Install Guide
Sourceforge Project

New GUI for Asterisk, Callware Voiceone.

Source: snapvoip.blogspot.com

A Source forge based Open Source project has developed an Management Gui for the IPPBX, Asterisk. I know all of you will say Go for Trixbox (I am already there ;)) but I want to give exposure to all applications that enhance Asterisk. I have not tried the product yet an all information I am providing is based on VoiceOne website.
The current release is 0.4.2. If you are a code Guru, you can grab the SVN version from the Sourceforge site.
From the outlook of the product, it seems promising. The interface is based on PHP and even though it is in it’s infancy many a features listed below are functional. They have an online demo that you could test out application and the function. Since these interfaces to Asterisk are personal tastes, some may prefer VoiceOne and asterisk to Trixbox!
Following are the feature list from the web site. So even if you already have Asterisk IPPBX management console, go check it out, give a boost to OSS developers.
* Client/Server architecture based on web services
* Relies on Asterisk real time Architecture (ARA) for database storage
* Two different panels, Personal for users and Configuration for administrators
* Extensions management
* Fully customizable users profile, including Voice mail, Call Forwarding ("Follow Me") and Do Not Disturb
* Highly configurable rule sets for outbound and inbound calls
* Static LCR (Least Cost Routing)
* Supports VoIP providers (SIP and IAX) and traditional Telco carriers
* Links remote offices via IAX with RSA public key encryption
* Powerful IVR creation system
* Queues management
* Conference rooms handling
* Sounds and Music On Hold management
* Applications and macros editor
* System Macros and Functions preloaded (DID/DDI, Call Back and DISA included)
* Plugins system to share ready-to-use macros and application with the VoiceOne community
* Powerful configuration of mISDN and Zap drivers based hardware
* Java SIP phone embedded
* I/O interface and PBX CLI (Command Line Interface)
* Static-like text editor for conf files

Links;
Callware VoiceOne
VoiceOne Demo
Quick Install Guide
Sourceforge Project

New GUI for Asterisk, Callware Voiceone.

Source: snapvoip.blogspot.com

A Source forge based Open Source project has developed an Management Gui for the IPPBX, Asterisk. I know all of you will say Go for Trixbox (I am already there ;)) but I want to give exposure to all applications that enhance Asterisk. I have not tried the product yet an all information I am providing is based on VoiceOne website.
The current release is 0.4.2. If you are a code Guru, you can grab the SVN version from the Sourceforge site.
From the outlook of the product, it seems promising. The interface is based on PHP and even though it is in it’s infancy many a features listed below are functional. They have an online demo that you could test out application and the function. Since these interfaces to Asterisk are personal tastes, some may prefer VoiceOne and asterisk to Trixbox!
Following are the feature list from the web site. So even if you already have Asterisk IPPBX management console, go check it out, give a boost to OSS developers.
* Client/Server architecture based on web services
* Relies on Asterisk real time Architecture (ARA) for database storage
* Two different panels, Personal for users and Configuration for administrators
* Extensions management
* Fully customizable users profile, including Voice mail, Call Forwarding ("Follow Me") and Do Not Disturb
* Highly configurable rule sets for outbound and inbound calls
* Static LCR (Least Cost Routing)
* Supports VoIP providers (SIP and IAX) and traditional Telco carriers
* Links remote offices via IAX with RSA public key encryption
* Powerful IVR creation system
* Queues management
* Conference rooms handling
* Sounds and Music On Hold management
* Applications and macros editor
* System Macros and Functions preloaded (DID/DDI, Call Back and DISA included)
* Plugins system to share ready-to-use macros and application with the VoiceOne community
* Powerful configuration of mISDN and Zap drivers based hardware
* Java SIP phone embedded
* I/O interface and PBX CLI (Command Line Interface)
* Static-like text editor for conf files

Links;
Callware VoiceOne
VoiceOne Demo
Quick Install Guide
Sourceforge Project

New GUI for Asterisk, Callware Voiceone.

Source: snapvoip.blogspot.com

A Source forge based Open Source project has developed an Management Gui for the IPPBX, Asterisk. I know all of you will say Go for Trixbox (I am already there ;)) but I want to give exposure to all applications that enhance Asterisk. I have not tried the product yet an all information I am providing is based on VoiceOne website.
The current release is 0.4.2. If you are a code Guru, you can grab the SVN version from the Sourceforge site.
From the outlook of the product, it seems promising. The interface is based on PHP and even though it is in it’s infancy many a features listed below are functional. They have an online demo that you could test out application and the function. Since these interfaces to Asterisk are personal tastes, some may prefer VoiceOne and asterisk to Trixbox!
Following are the feature list from the web site. So even if you already have Asterisk IPPBX management console, go check it out, give a boost to OSS developers.
* Client/Server architecture based on web services
* Relies on Asterisk real time Architecture (ARA) for database storage
* Two different panels, Personal for users and Configuration for administrators
* Extensions management
* Fully customizable users profile, including Voice mail, Call Forwarding ("Follow Me") and Do Not Disturb
* Highly configurable rule sets for outbound and inbound calls
* Static LCR (Least Cost Routing)
* Supports VoIP providers (SIP and IAX) and traditional Telco carriers
* Links remote offices via IAX with RSA public key encryption
* Powerful IVR creation system
* Queues management
* Conference rooms handling
* Sounds and Music On Hold management
* Applications and macros editor
* System Macros and Functions preloaded (DID/DDI, Call Back and DISA included)
* Plugins system to share ready-to-use macros and application with the VoiceOne community
* Powerful configuration of mISDN and Zap drivers based hardware
* Java SIP phone embedded
* I/O interface and PBX CLI (Command Line Interface)
* Static-like text editor for conf files

Links;
Callware VoiceOne
VoiceOne Demo
Quick Install Guide
Sourceforge Project

Asterisk + Outlook = Outcall

Source: snapvoip.blogspot.com

Bicom Systems, which is a provider of PBX and soft switch turn key solutions, has released it’s application OUTCALL, an outlook integration solution for Asterisk.
Bicom Systems announced today it has released its first freeware software to the “Asterisk Community”, OutCall. This is to be the first of similar releases of proprietary tools that can assist users with getting the most out of Asterisk and will also be released as freeware.

“Bicom Systems uses a variety of closed and open source software in its telephony systems. OutCall is an easy to use and install desktop application that assists users to integrate Microsoft Outlook™.” with use of making/receiving phone calls. OutCall was built for the purpose of working with PBXware that is Bicom Systems’ turnkey IPPBX and best results are to be had with PBXware. Nonetheless OutCall will work perfectly well with any Asterisk based system and integrate that system with Microsoft Outlook™.” said Stephen Wingfield, Bicom Systems.

OutCALL application is designed for integration with MS Outlook giving users powerful tools at hand placing and receiving calls. OutCALL features are:

Integration with one or unlimited system extensions (SIP/IAX)
Automatic integration with Microsoft Outlook 2000 and higher
Call History
Real time call notifications via pop windows
Placing calls within Outlook, email message or contact
Automatic contacts data update
Automatic application updates notifications
Clear debug information
Full PBXware / SWITCHware or vanilla asterisk integration
Also Developer/partner editions available

Links;
Asterisk outlook integration OutCall

Published on November 15th, 2006 under , , , , , , , ,

Asterisk + Outlook = Outcall

Source: snapvoip.blogspot.com

Bicom Systems, which is a provider of PBX and soft switch turn key solutions, has released it’s application OUTCALL, an outlook integration solution for Asterisk.
Bicom Systems announced today it has released its first freeware software to the “Asterisk Community”, OutCall. This is to be the first of similar releases of proprietary tools that can assist users with getting the most out of Asterisk and will also be released as freeware.

“Bicom Systems uses a variety of closed and open source software in its telephony systems. OutCall is an easy to use and install desktop application that assists users to integrate Microsoft Outlook™.” with use of making/receiving phone calls. OutCall was built for the purpose of working with PBXware that is Bicom Systems’ turnkey IPPBX and best results are to be had with PBXware. Nonetheless OutCall will work perfectly well with any Asterisk based system and integrate that system with Microsoft Outlook™.” said Stephen Wingfield, Bicom Systems.

OutCALL application is designed for integration with MS Outlook giving users powerful tools at hand placing and receiving calls. OutCALL features are:

Integration with one or unlimited system extensions (SIP/IAX)
Automatic integration with Microsoft Outlook 2000 and higher
Call History
Real time call notifications via pop windows
Placing calls within Outlook, email message or contact
Automatic contacts data update
Automatic application updates notifications
Clear debug information
Full PBXware / SWITCHware or vanilla asterisk integration
Also Developer/partner editions available

Links;
Asterisk outlook integration OutCall

Published on November 15th, 2006 under , , , , ,

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