Asterisk Software
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Note: Matt Florell sent this in for the Asterisk community. Thanks
We've released another update to our astGUIclient suite: 2.0.1
http://astguiclient.sf.net/
The client suite runs on most modern web browsers on almost any GUI-capable operating system, and it includes the astGUIclient client-side web app which extends your phone's functionality and the
VICIDIAL call center suite.
This package is free and GPL. (the suite is not an asterisk configuration tool) This package is geared towards Asterisk installations with SIP,IAX or Zap phones and Zaptel, IAX or SIP trunks.
For this major revision, we have reworked a lot of the server-side processes to make them more efficient as well as writing a new interactive installation script which uses a new central configuration
file. Another major new feature is the addition of a more advanced Predictive dialing algorithm for the outbound dialer portion of the system. We have also tested the suite on Asterisk versions through 1.2.12.1(cannot use 1.2.11 or 1.2.12 because of Asterisk bugs)
All client web-apps and administration pages are available in English, Spanish, Greek and German, with rough translations of French, Polish, Italian, Portuguese and Brazillian Portuguese for the client web-apps only.
Check out the project blog for more information:
http://astguiclient.blogspot.com
Let me know what you think.
Thanks,
MATT---
Written by Dal on September 18th, 2006 with no comments.
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LumenVox, an innovator of speech recognition technology, announced today that Digium Inc., the Asterisk company, now offers a fully-enabled Speech Starter Kit included in the Asterisk Business Edition at no additional charge. The Speech Starter Kit is also available to the Asterisk open source community for only $245.
“To enable widespread adoption of speech recognition solutions, we had to think differently. Through innovative collaboration with Digium, users can now easily develop Asterisk-based speech solutions and proliferate the market,” said Gerd Graumann, director of business development at LumenVox.
“This directly addresses the need for an affordable speech solution for the Asterisk community, and provides a significant additional benefit for Asterisk Business Edition customers,”
said Jim Webster, director of software technologies at Digium. This is the first time that Digium has offered a fully integrated commercial speech recognition technology for Asterisk. To ensure seamless integration, Digium created a unique connector bridge to the LumenVox Speech Engine.
The Speech Starter Kit includes the connector bridge and one LumenVox Speech Engine port, and can be deployed directly on any supported Asterisk server. Asterisk users can purchase the Speech Starter Kit directly from Digium's website.
Test-drive the LumenVox speech recognition software at the upcoming VON show in Boston (www.von.com), at www.LumenVox.com or call 877-977-0707 and say “Demo.”
Written by Dal on September 12th, 2006 with no comments.
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Here are the main features:
- Simpler build system, no messy patching!
- CDR record handling should work correctly now
- Will set the DIALSTATUS variable
- Change pitch during conversation with * and #
- Voice effect can be applied to peer channel instead with 'p' option
The simpler build system is partially due to help from anthm for wrapping my modifications to SoundTouch in to a separate library found here: http://www.lobstertech.com/code/libsoundtouch4c/
Please note that this version is a total rewrite and I haven't had a whole lot of time to valgrind and test it rigorously. If you have ANY problems, even if you figure out how to work around them, please let me know so I can fix it so others don't have the same problem. I usually respond pretty quick to email (in the 5 minutes to one day range)
Plans for 0.6:
- Change voice pitch via manager api and command line
- Open to suggestions
Have fun!
- Justin Tunney
Written by Dal on August 25th, 2006 with no comments.
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Hello everyone,
I have released AstLinux 0.4.3. For all of those that have been waiting to switch to 0.4.x, this is your chance. The few remaining problems with uclibc have been fixed (i.e. voicemail timezones and voicemail -> email via MSMTP).
Don't forget to peek around in SVN for all kinds of goodies. Especially trunk - the Gumstix is now a direct target for builds. That's right, build AstLinux for a Gumstix just as easily as a Soekris!
--
Kristian Kielhofner
Download:
http://sourceforge.net/projects/astlinux/
Written by Dal on August 24th, 2006 with no comments.
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Calling Circles Desktop (Asterisk Edition) v 1.0.6 has been released. Please make sure you uninstall prior versions (including release candidates before installing)!
Summary of improvements in version 1.0.6:
* Smart Tags enable dialing from Microsoft Office applications. Make sure Smart Tags are enabled in your office applications to take advantage of this new feature.
* Dialing in Internet Explorer is enabled. Highlight a phone number on a web page, right click, and select Calling Circles Dial.
* A new protocol will be added to your computer called "phone:" You can dial a phone number from the Windows Start menu run dialog. Select Start->Run and type phone: where is the phone number you'd like to dial. Now you can integrate dialing from any application without the need to learn COM!
All of the Calling Circles Desktop dialing rules will be applied when using any of the above methods to make a call. Thank you to all of our early adopters who provided fantastic feedback! To download this release, click here. If you need a new evaluation key, please send me an e-mail! elliot.otchet@callingcircles.com
Click Here for more Information
Written by Dal on August 23rd, 2006 with no comments.
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Hello,
We are releasing a beta version of our java eclipse based asterisk voicemail client called "Tycho". You can download the client binary at: http://sip-syndication.com.
Please note that the client needs a set of php scripts installed on the server side. The server side scripts, called "vmxml" are open source and available on the same website.
Screenshots:
MS Windows
http://sip-syndication.com/images/stories/tycho/msw-tycho-overview-65.png
Mac OS X
http://sip-syndication.com/images/stories/tycho/tycho_macosx_ppc_overview-65.png
Linux
http://sip-syndication.com/images/stories/tycho/lnx-ubuntu-tycho-overview-65.png
If you have any questions or problems installing/using the client please mail: t y c h o at sip-syndication dot com or post to the support forum on our website.
cheers,
Arnd
Written by Dal on August 21st, 2006 with no comments.
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I am pleased to announce a new release of Asterisk2Billing V1.2.3 (
BrainCoral)
http://asterisk2billing.org
NEW FEATURES HAVE BEEN ADDED, YOU WILL FIND THE DETAILS BELOW:
- New Reporting for Earnings : Viewing of Profits, Sells, purchase...
- English correction
- Better support for Internet Explorer
- Signup mail activation
- on Customer interface, new "forget password" feature
- DID import tool
- CARD import tool
- Voip Provider & trunk filtering in call reports
- Add markup & margin on CDR report
- Delete option after selection into card & rates browsing
- New recurring service "auto refill"
- New languages Russian & Portuguese
Project page: http://asterisk2billing.org
WIKI -> http://wiki.asterisk2billing.org/
FORUM -> http://forum.asterisk2billing.org/
DEMO -> http://demo.asterisk2billing.org/
Kind regards,
/Areski
Written by Dal on August 13th, 2006 with no comments.
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ESCAUX net.PBX is a turnkey Asterisk solution designed for the SME and Corporate customer. We made Asterisk easy. Expercience for yourself and download the ESCAUX net.PBX Free Edition today.
The software is ready for download at: http://www.escaux.com/netpbx
ESCAUX net.PBX runs directly from CD, requires no installation and does not overwrite your harddisk. Your configuration changes are centrally stored on our servers and become instantly available at every system reboot. Simply place our Smart Live CD in your PC, follow the Configuration Wizard and show-off to friends and colleagues with your "Web controlled, Asterisk based, Business IP PBX".
The net.PBX Free Edition is a full featured IP PBX system suitable for Business use. You can install and run the Free Edition on your own servers and benefit from a close to zero cost IP PBX system for your company. On Demand commercial support packages are available to assist you with installation or configuration but you can also refer to our net.PBX Community Forum.
The ESCAUX Free Edition is an Open Source project hosted at SourceForge.
At any moment in time you can decide to migrate towards a Commercial version of our product and benefit from our unmatched Service Level Agreement, Guaranteed response and Repair times.
Best regards,
the ESCAUX development team
Written by Dal on August 9th, 2006 with no comments.
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Hello,
We've just released our Libero Management System application, a web-based interface to configure and manage your Asterisk-based PABX. Designed for the not-so-novice Asterisk administrator in mind.
LMS is simple to install, has minimal requirements (no external databases or components required), and runs on Apache and PHP.
Some features:
* Point-and-click dialplan configuration
* Javascript-based real-time PABX monitoring tool
* configure Zap/IAX/SIP channels/trunks/extensions
* Zaptel interface configuration
* CDR reporting
* Sound file and MOH management.
A 30-day evaluation version is available. For more details, please see:
http://www.lanvik-icu.com/lms/index.php
Thank you
Written by Dal on July 31st, 2006 with no comments.
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Hi All,
For all those using asterisk + voismart gsm cards, we have released a new package that fixes a lot of issue and add some new features.
Take a look to voismart open source website:http://open.voismart.itGreetings,
Matteo.
Written by Dal on January 1st, 1970 with no comments.
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Packet Island today announced general availability of ‘PacketSmart for Asterisk’, a complete VoIP lifecycle management solution for the open source Asterisk PBX. Packet Island’s ‘PacketSmart for Asterisk’ is a Software-as-a-Service (SaaS) solution that has been designed to target the VoIP lifecycle management needs of VARs and SMBs deploying Asterisk.
The solution is based on Asterisk software agents and purpose-built micro-appliances that work with the highly scalable PacketSmart SaaS platform, hosted at a Tier-1 data center. The Asterisk software agents which are installed on the Asterisk PBX perform continuous monitoring of all VoIP calls terminating at the PBX. The 4”x4” micro-appliances are complementary to the software agents, in that unlike the software agents, they can be easily moved around the network to isolate and troubleshoot network issues affecting VoIP quality.
The “PacketSmart for Asterisk” solution has three distinct functions:
* Network assessment for VoIP
* VoIP troubleshooting
* Ongoing VoIP SLA monitoring
During the deployment planning phase, the micro-appliances can be used to simulate live VoIP traffic on the SMB network, to identify and fix problems before the Asterisk VoIP deployment. After Asterisk deployment, the Packet Island software agent that is left installed on the Asterisk PBX continues to collect detailed quality metrics that can be used by the SME and VAR for SLA monitoring and to isolate and troubleshoot transient VoIP quality issues. For multi-site deployments with multiple Asterisk PBXs, the software agents can be employed to do periodic network assessments to ensure that the inter-Asterisk connectivity continues to support good VoIP quality.
Netswitch, a leading technology management company based in San Francisco, has been using the ‘PacketSmart for Asterisk’ solution for Asterisk deployment planning at the US offices of a large multi-national company. “We looked at several VoIP management solutions in the market, but none of them matched the flexibility and ease-of-use we found in Packet Island’s solution. The micro-appliances are truly plug-and-play and can be easily moved around in the network to isolate congestion points and evaluate routes. Furthermore, with the software agents installed on the Asterisk PBX, we can now offer our customer a continuous monitoring service that is essential in staying on top of transient congestion issues that affect VoIP” said Stanley Li, CEO of Netswitch.
“Open source Asterisk is creating a huge disruption in the PBX marketplace. However, one of the big challenges in deploying Asterisk with VoIP today is in ensuring that the customer’s network is ready for VoIP before the Asterisk deployment, and to stay on top of transient VoIP quality issues that invariably crop up as the network evolves.
The problem with current VoIP assessment and monitoring solutions in the market is that they are more expensive than the entire Asterisk PBX. With ‘PacketSmart for Asterisk’ we are offering the industry’s first SaaS solution that has been specifically created for the VoIP lifecycle management needs of the Asterisk market. Instead of paying thousands of dollars up front, our customers pay us an annual service fee that works out to tens of dollars a month“ said Praveen Kumar, President of Packet Island Inc.
Written by Dal on January 1st, 1970 with no comments.
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Note: If you have not looked into this software for you high volume or call center pbx, this is a great application. In my consulting we have deployed this on quite a few installations. Glad to see the wallboard kioks functionality added.
We are pleased to tell you that we have released a new version of QueueMetrics. The main areas of improvement were the following ones:
- Internationalization engine: the text of QueueMetrics can be easily localized, as well as numbers and dates.
- Internationalized versions: Italian, German and French
- Export all data to Excel, CSV files or XML
- Database update utility: makes the database up-to-date to the latest version of QM.
- You can now connect to an Asterisk server for call barge-in by using the Manager interface as well as call-files.
- You can now create a single-URL login and show realtime screens for kiosks or wallboards.
A full list of improvements over version 1.2.1 can be found at:
http://queuemetrics.loway.it/news.jsp
QueueMetrics 1.3 allows data storage on both flat files and MySQL databases for bigger call centers. And of course comes with a 90-page user manual that covers all aspects of it.
QueueMetrics is a commercial call center monitoring package, but is availabe free of charge for individuals, Asterisk hackers and small SOHOs. You can request a trial key if you run a larger installation and would like to test it in your own environment.
The latest version of QueueMetrics can be downloaded from:
http://queuemetrics.loway.it/download.jsp
Hope you like it,
l.
Written by Dal on January 1st, 1970 with no comments.
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Mexuar Communications is today launching the Corraleta Technology SDK, an innovative new solution that enables rapid development and deployment of VoIP click-to-talk applications for online businesses. The Corraleta SDK enables deployment of click-to-talk functionality on the website of any size of business.
With this feature, any website visitor can use their web browser and PC to make free calls and talk with sales or support staff in the company’s contact centre. Research from several leading industry analysts has shown that click-to-talk solutions increase online sales, by boosting browse-to-buy conversion rates and improving customer service.
The Corraleta Technology SDK takes users from a web page to a VoIP call with a live customer service agent within 10 seconds. These calls are pure IP from the web browser to the customer contact centre PBX and are free, irrespective of the location of the caller or contact centre.
The Corraleta-developed application can be used to initiate calls to the contact centre via a button on a web page, or can trigger a callback from the contact centre to the user’s chosen telephone. Application examples include sales and support services for online merchants; emails embedded with click to call functionality; and pay-per-call online banner advertising
The Corraleta Technology SDK uses Java to deliver VoIP across multiple browsers and 3rd party white label applications, providing tight, platform-agnostic integration between traditional phone systems and core business applications, whether on the company internal network or on the Web. This contrasts with rival solutions that use ActiveX and work only with Windows platforms and the Internet Explorer browser.
According to Mexuar’s director Tim Panton: “The ability for web users to interact directly with customer service agents is proven to boost sales and overall service – especially in higher-value sales transactions. The Corraleta Technology SDK delivers click-to-talk 2.0 for businesses, enabling them to quickly deploy platform-neutral applications to streamline communications, enhance sales and service, and deliver communications cost savings.”
In addition to enabling IP calls, the Corraleta SDK can also delivers information about the user’s online experience to the merchant’s customer service agent when the call is initiated. The agent software can display this information directly or use it to trigger lookups in company databases to retrieve details such as customer records, purchase histories and so on.
At Astricon Dallas (www.astricon.net), Mexuar director Tim Panton will be giving presentations about the technology behind Corraleta and its ability to integrate Asterisk IP PBXs and web applications on Wednesday 25th Oct at 16:30; Thursday 26th Oct at 14:00 and Friday 27th Oct at 12:30.
Written by Dal on January 1st, 1970 with no comments.
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Note: After going over the admin interface, I think its pretty user friendly so far. I think I might give the a through run down to see what trade offs you are making when user the management interface.
We've released VoiceOne 0.4.0, a web-based and open source solution which allows to fully manage an Asterisk service hosted on a LAMP server. We focused on an charming and overall user-friendly interface. Thanks to the authentication based on roles, once configured by a super user, the PBX may be easily maintained even by an Asterisk unskilled users.
From a technical point of view, the application is made up of two modules: one for the client - i.e. the user interface - and the other for the server. Thanks to the web services provided by the server module and the use of a database, VoiceOne may be easily integrated with other applications (e.g. CRM software).
The project has grown and has received positive response so far. Nowadays there's a little but enthusiastic community of developers, supporters and users. Translations in several languages (e.g. English, Spanish, Russian, etc.) are already available.
On the project website at you'll find the online demo and the links to download the source files from Sourceforge, as well as a support forum. We would be pleased if you could give it a try and let us know your feedback, comments, ideas, or suggestions replying here or posting a message on our forum. Thanks for your kind attention.
Regards,
Alex
Written by Dal on January 1st, 1970 with no comments.
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Note: Sean from i9 sent this over to me and looks interesting. Below I have pulled a little info from there site.
It's my pleasure to introduce you to a new product for the open source project Asterisk. iSymphony is a cross-platform, real-time operator panel for Asterisk that comes in two flavors. A free downloadable version and iSymphony Conductor Edition.
iSymphony, an easy-to-use, Java-based client/server software for managing phone calls via i9's iPBX or the Open Source Asterisk platform. Due to Java's architecture, our software runs on many different computer operating systems and its user interface is clear and easy to master. download iSymphony displays:
* What extensions are busy, ringing or available
* Who is talking and to whom * Registration status and reachability
* Parked Calls Features:
* Hang-up a channel
* Transfer a call via drag & drop
* Originate calls via drag & drop
* Remotely monitor calls in real time from your PC (barge)
* Record calls * Mute/Unmute barged calls
* Per user authentication
At a quick glance, you will be able to see incoming calls and calls on-hold as well as the availability of all organizational personnel.
More Information Here:
www.i9technologies.com/isymphony
Written by Dal on January 1st, 1970 with no comments.
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Hello everyone,
I have released AstLinux 0.4.4. Thanks to all of the testers on astlinux-users, AstLinux 0.4.4 now includes mISDN support (
again). Grab AstLinux at
http://www.astlinux.org.
--
Kristian Kielhofner
Written by Dal on January 1st, 1970 with no comments.
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VoIP and VVoIP (
Voice & Video over IP) solutions provider
Adiance Technologies launches Alpha Version of 1Videoconference. One of the World’s First open source web based video conferencing solution for
Asterisk. Developers at Adiance have again succeeded in becoming some of the world's first innovators to stream asterisk audio/ video channels to web browsers, clearing the way for the most advanced video solutions such as, IP TV, video conferencing, video contact center, remote surveillance, video kiosk and video pbx etc.
1VideoConference is an open source solution uploaded on Source forge. Released recently, this product allows web, audio/ video phone, skype, msn and yahoo users to participate in multipoint video conference. It empowers businesses to offer innovative online video services like TV shows, photography shows, marketing events, network games, social community sites, video wiki, blogs and forums, video tutorials, video library and more.
1VideoConference has turned out to be a great boon in Disguise. A farmer can now show his destroyed crops to Scientist sitting in another corner of world & know the exact reason of harvest failure within few minutes. World best learning Institutes Oxfords & Harvard’s can now impart education at a time to many students at a time spread across the world. Companies spread across the World can have board meetings together at a time thus saving cost as well deciding new strategies. 1VideoConference has truly turned as a cost & time saving boon for community at large. Blending the advanced technologies of .NET3.0 & Asterisk it has brought the World all the more closer.
1videoConference is tailor-made according to the present and future need of the customers. One of the extensive features of it is that if the internet connection of any of the meeting attendees goes down then with the help of coordinator they can switch over to the mobile and can continue at least with voice conferencing.
1Videoconference is a rich web communication system that gives professionals from all spheres of the world full control of enterprise-class web conferencing, online training, Audience poll, Destop-Sharing,co-browsing,Whiteboard, weblink, remote monitoring, co-authoring & many more features for collaborative communications of entire organization. All one needs is a desktop, a headphone, moderate internet connection of 256kbps. Combination of open source Asterisk software and state of the art Adiance platform enables customers to implement VoIP video solutions at more than 40% cost savings as compared to available solutions in the market. 1Videoconference is available on both a hosted basis and as a software that can be deployed on-premise.
In the words of Mr. Hardik Sanghvi President and CEO of Adiance, "1videoConference is true enabler of video/ voice and data interactions among the web and phone users." Sanghvi went on to say, "Being open source, webbased and foot print free, video enabled and cost effective, 1videoConference, as I envision, will change the way people and businesses collaborate on day to day basis ."
Source: Adiance Technologies
Written by Dal on January 1st, 1970 with no comments.
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Hi all!!
I've made some changes to the applications that Astertest was using to monitor the performance of the server. Now is also possible to track the bandwidth usage of the server, this has nothing to do with the executable (astertest.exe) itself but with the events that the Asterisk Manager generates.
The method described in:http://www.asteriskguru.com/tutorials/astertest.htmlto perform the test is still valid.
In the next days I am gonna make available some scripts to originate the calls and to make some graphs of the test, just like astertest does ;)
You can find the source here:http://toofic.no-ip.org/pub/src/app_securax.tar.gzI've compiled them against Asterisk 1.2.12.1, but I think there should not be problems with other versions.
I hope someone could find it useful.
--
Grettings,
Víctor Toofic
Written by Dal on January 1st, 1970 with no comments.
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At last!, I have finally finish the Windows GlassConsole. A fully functional windows console for the Asterisk PBX. People have always askme for an easy to use hand off linux, Asterisk Console. To be totally graphicall and not having to deal with the struggle of going trough a web site or a linux prompt. Well, here it is!.
GlassConsole, is a message driven console that allows you to monitor every single aspect of your Asterisk server. From active calls, conference rooms, agents , queues, channels. Not only that it gives you alarms triggers. Where you can setup alarms in real time based on events or alarms on the systems. These can be either visually on the screen , to email or to file. Everysingle alarm is also autoamticly log to a file. For those that use asterisk in a production environment Glassconsole is amust monitoring tool to have.
Features:
With the GlassConsole you can view:
- Actual bridge calls.
- Actual register phones in the system.
- Number of CHannels being use either in Zap, Sip, Iax.
- Number of consference rooms and people in it.
- Number of Agents logged in or out and time being in Queue.
- Status on call Queues.
The single server license has the following alarms:
- Alarms if the console is exited.
- Alarm when the console get s started.
- Alarm if connected to the Asterisk server.
- Alarm if you loose connectivity to the server.
- Alarm if a phone does not re-register in the time required.
- Alarm if a Zap Channels goes down.
- Alarm when a Zap Channels clears.
The following alarm are only available in the enterprise:
- Alarm on maximun Zap channels.
- Alarm on maximum Sip channels.
- Alarm on maximum iax channels.
- Alarm on minimun Agents logged in.
Requirements:
The software requirements for GlassConsole V1. are as follows:
The screen resolution needs to be 1280 x1024. Recommended a 19" monitor. Reason: There is a lot of information to be display.
Windows XP or 2005
1GB of memory RAM, more might be need it if the Asterisk Server is really busy. (A lot of simultaneous calls, or to many registered phones).
An Asterisk Server running and configure to receive Remote management connection.
Click For More Information and Demo Download
Written by Dal on January 1st, 1970 with no comments.
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Celliax is a channel driver for the Asterisk Free PBX that manages GSM and CDMA cellular phones through an adapter, composed by a datacable (for commands) and an audiocable (for the voice) interfacing the computer soundcard. Runs on Linux and Windows (
click below for more).
What's New in 0.0.32:
* LiveCD:
o Now you can save all your configuration between reboots
(Celliax, Skype, Language, etc). You can save all of them on a regular file on your hard disk (or on a USB key) and have all of it automatically reloaded at next boot. No more need to "retrain" Skype
at each reboot!
o Linux support for all kind of datacables in the Celliax Devkits
o New kernel 2.6.17.14 with ark3116 and working wireless
ipw2200 modules
o Headless Celliax, cellphones, Asterisk and Skype (without X)! Booting in console mode you find a script
(/usr/local/asterisk/usr/sbin/celliax_server) that automatically launch Xvfb (the fake X11 server), Skype and Celliax. Also via ssh! Isn't that cute?
o Graphic configuration for Celliax
o Automatically download the latest sources with an svn script (/usr/local/asterisk/usr/sbin/svn_celliax, now's working)
o Contains the Windows XP drivers for the Celliax DevKit datacables
* Celliax (both Linux and Windows):
o Asterisk 1.2.13
o New configuration file format
o Better interaction with Skype (no more stuck in skype_dir, much faster skype_dir on Windows)
o Source code is more readable (but will be much cleaned, refactored and polished in the future ;) ) and beginning of Doxygen documentation
* And many more minor things...
Celliax development and download site: www.celliax.org
Written by Dal on January 1st, 1970 with no comments.
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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
- Developer/partner editions available
Click Here for more Information
Written by Dal on January 1st, 1970 with no comments.
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Asterisk IT have released beta 3 of AsterFax 1.1. Beta 3 is likely to be the last beta before AsterFax 1.1 is released. The key feature of AsterFax 1.1 is that in conjuction with SpanDSP 0.3 Asterisk finally has a reliable Faxing service which will operate on all Digium hardware for both inbound outbound faxing.
Asterisk IT have released beta 3 of AsterFax 1.1. Beta 3 is likely to be the last beta before AsterFax 1.1 is released. The key feature of AsterFax 1.1 is that in conjuction with SpanDSP 0.3 Asterisk finally has a reliable Faxing service which will operate on all Digium hardware for both inbound outbound faxing.
It should be noted that the SpanDSP libraries still have some compatability problems with a small number of fax machines. For 100% compatiability with all fax devices Asterisk IT recommends using AsterFax with a hardware based fax system such as the Eicon Diva range of boards or any linux compatible TTY fax card. AsterFax provides an Email to Fax gateway for Asterisk. AsterFax lets you send an email by Fax.
Enter the phone no. in the 'To' address, compose your email message and click send. You can also fax an attachment or directly from Word and OpenOffice. AsterFax also provides Unified messaging by delivering inbound faxes directly to a users mailbox. With AsterFax each staff member of an organization can have their own private fax number.
You can download the beta from
here. A single fax line version of
AsterFax is available freely.
If you wish to purchase a commercial license for us with more than a single fax line then details are available from the
purchasing page.
Written by Dal on January 1st, 1970 with no comments.
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Note: Matt Florell posted about the new release of the astGUI Client
astGUI 2.0.2 client suite runs on most modern web browsers on almost any GUI-capable operating system, and it includes the astGUIclient client-side web app which extends your phone's functionality and the VICIDIAL call center suite. This package is free and GPL. (the suite is not an asterisk configuration tool) This package is geared towards Asterisk installations with SIP,IAX or Zap phones and Zaptel, IAX or SIP trunks.
For this release, we have made many changes on the server side, including changing logging from AGI to FastAGI which can cut system load in half on busy VICIDIAL systems. We have also tested the suite on Asterisk versions through 1.2.14 (cannot use 1.2.11 or 1.2.12 because of Asterisk bugs).
All client web-apps and administration pages are available in English, Spanish, Greek and German, with translations of French, Polish, Italian, Portuguese and Brazillian Portuguese for the client web-apps only.
Check out the project blog for more information:
http://astguiclient.blogspot.com
Click Here for Download:
http://astguiclient.sf.net/
Written by Dal on January 1st, 1970 with no comments.
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Note: Nerd Vittles does it again. If Asterisk help articles wasn't enough, now he is dropping software on us. What's next, book or lecture series???
Nerd Vittles today introduces a free Windows utility that was our most requested application in 2006...
Excerpt: It's been five days since U-Rang hit the street and, with over 1,000 downloads, it seemed like a good time for an update. After all, it's a new year so out with the old, and in with the new. Version 2 introduces an entirely new user interface and an intuitive calling log based upon (what else) phone slips, two of the most requested additions from the early adopters.
When an incoming call arrives, the ubiquitous phone slip will pop up on your desktop with the time and date of the call as well as CallerIDname and number information of the caller. And, if you're using the Nerd Vittles CallerID enhancements, then your Asterisk system will check the Google Phonebook, AnyWho, and AsteriDex for supplemental CallerID information in addition to what's provided by your local phone company.
Article: http://nerdvittles.com/index.php?p=162
Written by Dal on January 1st, 1970 with no comments.
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We've been holding back on this release to coincide with the Asterisk 1.4.0 release.
This is mostly a compatibility release, but there are a few new features:
* No longer requires register_globals in PHP
* Separated code from configuration settings in ./lib/defines.php (hopefully this will make future upgrades easier)
* Migrated all database interfaces to PEAR::DB which simplifies the code a bit and opens
up the possibility of using other databases to host the scheduling DB (app_cbmysql is still only MySQL, but ODBC is planned/hoped for)
* The conference monitoring code now uses the concise output from "meetme list", improving
the parsing of participant details.
* Minor tweaks to improve the cbEnd.php script that enforces the conference duration, plays announcements and populates the conferencing CDRs.
* Conference CDR records now store participant duration in seconds instead of a formatted string, allowing for further analysis (the web interface still formats the duration for display purposes)
* App_cbmysql is updated to work with Asterisk 1.4.0
* App_cbmysql has it's own build environment now, no longer requiring a Makefile patch, etc...
The new release can be found at: http://sourceforge.net/projects/web-meetme/We do have a volunteer developer who will be maintaining the 2.X.X chain for Asterisk 1.2.X compatibility, so bug fixes and features that are not Asterisk version dependant will still be made available for older installations.
Thanks,The Web-MeetMe development team...
Written by Dal on January 1st, 1970 with no comments.
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Note: Found this on the list this morning. Looks promising after reading the description for those integrators that still use xp/2000 as the daily desktop.
Originally an exercise to learn Asterisk and have a GUI of my own to use, I developed a Windows based GUI to build dial plans and upload them to the Asterisk server. Currently in beta, it's aim is to abstract routine chores such as dialing an extension or playing a voice.
I also wanted to be able to implement custom code in a easy graphical way as well so I included a scripting editor with most of the core functionality you'd expect like syntax highlighting, Parameter Hints, etc.
Although the GUI is Windows based, it communicates with a Linux binary TCP socket server written in house to control basic Asterisk functionality such as uploading required or included files, issuing simple commands like reload, restart, etc over the network. I also have plans to write a function to remote debug an AEL script using the aelparse executable and it's output sent back to the Windows GUI.
While definitely still in beta, we are using the software to program our own Asterisk box here in our office and it's working very well for us. Although note that we have a fairly simple dialplan with just a little bit of conditional logic, FirebirdSQL access and some TTS stuff to tease our resellers when they call in ;).
We develop and sell a different product, but Asterisk PBX systems compliment our other products very nicely and we already have a very strong reseller base. So we'll be selling this software ourselves to local end users and to our own reseller chain.
On the other hand, I'm not quite sure how it would fit in the Asterisk community at large, if at all. It's not an end to end, black box GUI for end users. There's already better ones out there already.
It's not a complete "developer" tool either so it's it hard to say if it could be useful to others outside our company.
I don't have any plans to spam the list, even the biz list, but wouldn't mind some feedback if anyone has a few minutes to take a look.
http://www.datatrakpos.com/pos/datatalk/Default.aspxThanks to everyone for help on the lists.
--
Warm Regards,
Lee
Written by Dal on January 1st, 1970 with no comments.
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Call Accounting Mate announced today that it has developed integration for the massively popular Asterisk the Open Source PBX into its
Call Accounting Mate software. The move is largely in response to the large number of inquiries from developers and integrators of Asterisk.
Call Accounting Mate now offers a tailored set of call logging components exclusively for the Asterisk product family. This new mechanism of interoperability provides seamless and dynamic IP integration for real time data collection and monitoring of
call detail records (
CDR). callaccounting.ws now offers special Javascript drivers that interpret the dynamic CDR data from Asterisk boxes into a standard format.
Customers can utilize Call Accounting Mate to enhance productivity, traffic analysis, corporate cost allocation, accounting integration, workforce management and customer billing.
For more information about Call Accounting Mate integration with Asterisk,
please visit the website forum:
http://www.callaccounting.ws/forum/index.php?board=5.0
Written by Dal on January 1st, 1970 with no comments.
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Note: DataTrak has released this API Utility. Email me if you put some use to this program and if its useful for you.
Written by Dal on January 1st, 1970 with no comments.
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