Asterisk VoIP News

Friday, September 30, 2005

Co-author of O'Reilly's Asterisk book presenting in Utah Valley

This post just cam across the User List about the Utah Valley Linux Users Group(UVLUG).

"On Saturday Oct 8, Jared Smith, co-author of O'Reilly's recent top-selling book "Asterisk: The Future of Telephony" will present to the Utah Valley Linux Users Group on Asterisk and VoIP.

Anyone in the area (Utah Valley) is welcome to join us (UVLUG) at this free event. Besides being a great presentation, there will be plenty-o-swag (books etc..)

If you need more info, this page has it:

Thursday, September 29, 2005

Packt Publishing Announces a New Book on Asterisk PBX

Building Telephone Systems With Asterisk is a revolutionary guide that will help bring Asterisk main stream. Asterisk runs on Linux, BSD and MacOSX and provides many of the same PBX features found in enterprise level pbx systems. "Asterisk does voice over IP in many protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk merges voice and data traffic seamlessly across disparate networks"

Who this book is written for?

This book is aimed at anyone who is interested in building a powerful telephony system using the free and open source application, Asterisk, without spending many thousands of dollars buying a commercial and often less flexible system. This book is suitable for the novice and those new to Asterisk and telephony. Telephony or Linux experience will be helpful, but not required.

You can download the sample chapter in our Press Release section.

This book takes you from installing and configuring Asterisk to using its various advanced features-helping you build feature-rich telephony systems.

About the Authors

David Gomillion currently serves as Director of Information Technology for the Eye Center of North Florida. There, he orchestrates all of the technological undertakings of this four-location medical practice, including computers, software (off-the-shelf and custom development), server systems, telephony, networking, as well as specialized diagnostic and treatment systems.
David received a Bachelor's of Science in Computer Science from Brigham Young University in August, 2005. There he learned the theory behind his computer experience, and became a much more efficient programmer.

Barrie Dempster was a Network Administrator/IT Manager for a growing call center when he saw the convergence and dependence of telephony and IT-related fields on each other. He focused on integration of telephony with IT infrastructure, and took on security as a career. The increase of voice-over-IP communications has now led to high demand for these skills, which he now utilizes in his current position as a Scotland-based Infrastructure and Security consultant for a variety of clients primarily within the financial sector.

In summary this "book shows how to build a telephony system for your home or business using the free and open source application, Asterisk. "Building a Telephony System with Asterisk" takes you step-by-step through the process of installing and configuring Asterisk."

Click Here to Preorder your copy Today

Tuesday, September 27, 2005

Announce: AstriCon 2005 - Now With Free Beer!

Steve Sokol has posted the lastest update for AstriCon 2005. We better see everyone there this year, lets make it one for the books.

AstriCon Update: Only Two Weeks To Go!
October 12 - 14, 2005
Anaheim, CA

AstriCon 2005 starts two weeks from today. We now have a complete roster of speakers covering Asterisk from soho to carrier. We've added the Code Zone, a working lab with a full compliment of VoIP and TDM equipment. We also have over 20 confirmed exhibitors and more are joining the event each day. AstriCon 2005 is shaping up to be a three day Asterisk extravaganza.

AstriCon 2005 Highlights:

- Keynotes from Asterisk Leaders:
* Mark Spencer: Asterisk 1.2 And Beyond
* Carrier Grade, Fault-Tolerant Asterisk
* Asterisk VoIP & Emergency Call Handling

- Asterisk 1.2: Enhancements, Features & Changes

- Free copy of "Asterisk: The Future of Telephony" from O'Reilly
* By Jim VanMeggalen, Leif Madsen, & Jared Smith
* Free to the first 500 tutorial attendees

- 3 SIG Tracks:
* Enterprise Asterisk
* Call Center Operators
* ITSPs & Carriers

- The Asterisk Expo: 20+ Asterisk Related Vendors

- The Open Source Showcase: Asterisk-related open source projects
* AstLinux * AsterNIC
* Asterisk on WRT54G * astGUIclient/VICIDIAL
* Zap Radio

- 3 Tutorial Tracks:
* Beginners
* Intermediate & Advanced
* Developer

- The Code Zone: a lab stocked with hardware and Red Bull
* Come in and code!
* Meet the gurus of Asterisk.
* Test out solutions
* Show off your code

- Huge Party: The Golden Asterisk Pub
* J.T. Schmidt's Brewery
* Free Beer (Thanks Digium!)

Register today:

Don't forget that space at AstriCon 2005 is limited. Please register as soon as possible to insure admission to your preferred tutorial and conference tracks. Hotel space is also limited.

Reserve your room today:

Please contact us if you have any questions: or by
phone at +1 816 256 8916.

Review: Digium 405P v2 Firmware Reviewed

This came across the User List post by Matt Florell. Thanks Matt!

"A couple months ago Digium announced their new firmware version for the TE4XXP series of T1 cards(TE405P/TE410P) as well as announcing the new dual port cards(TE205P/TE211P) and the echo-canceller cards(TE406P/TE411P). After several weeks of asking for a test TE406P card, I was able to get Digium to put an echo-canceller daughter card on to a TE405P, that I was sending in for a firmware upgrade anyway, so that I could test it for a couple weeks."

Click Here for the Full Article

Press Release: SANGOMA TECHNOLOGIES Registers Patent Application on Echo Detection And Control System

Technology to improve performance of host-based HMP and hardware-based telephony systems

TORONTO - September 27, 2005 – Sangoma Technologies Corporation (TSXV: STC) (, a leading provider of connectivity hardware and software products for VoIP, TDM voice, WANs and Internet infrastructure, today announced that it has filed a patent application for a method to enhance the efficiency of echo cancellation hardware and software. The provisional patent application titled "Echo Canceller Controller" covers the techniques of measuring echo on incoming voice streams and controlling the echo canceller appropriately.

"This technology is a major step forward in improving the performance of echo cancellation, particularly in the Host Media Processor (HMP) environment," says Sangoma Technologies President and CEO David Mandelstam. "The trend for PC-based telephony is away from expensive hardware solutions toward HMP implementations such as NetStructure by Intel or the open source Asterisk telephony system. High quality echo cancellation on HMP systems consumes approximately 30MHz of host CPU overhead per telephone conversation, so there is considerable scope for reduction of CPU load in systems supporting hundreds of simultaneous calls."

Sangoma's EDAC (Echo Detection and Control) is an algorithm that examines each call as it is connected, and within about one second, determines whether the call has echo or not. It then enables or disables the echo canceller as necessary. Because all long distance, cell phone, ISDN and VoIP calls are either already echo cancelled or have no echo, echo cancellation loads on HMP systems can often be reduced by 80 per cent or more. Where echo cancellation is hardware-based, EDAC can reduce the up-front cost of echo cancellation hardware to match the mix of local analog and other calls that are expected on the system.

"EDAC will be implemented as a standard application in our on-board firmware at no cost," adds Mandelstam. "We expect EDAC to become a must- have feature in the delivery of practical HMP and low- cost hybrid HMP/hardware-based VoIP gateways, PBX and IVR systems and other telephony systems."

Monday, September 26, 2005

Announce: AstBill Released!

What is AstBill?

AstBill is open source software licensed under the GPL, and is maintained and developed by a community of users and developers. AstBill is free to download and use. If you like what AstBill can do for you, please work with us to expand and refine AstBill to suit your needs.

AstBill is not only a web-based, user-friendly billing interface to Asterisk. It is also a Asterisk configuration and management tool and a standardized implementation of Asterisk using REALTIME and static configuration as you please.

Here are some of the features of AstBill:

User-friendly End User Web interface gives access to a range of functionality:

-Personal Contact Directory with Categories
-View SIP, IAX and Virtual Accounts
-Virtual Accounts (Lets you forward your calls to any extension you want also time based forwarding)
-Credit Control on outgoing calls
-Show Balance, Expenditure, Payments and number of Calls on each account
-Set warning balance for email when low credit on account
-View Numbers Dialed and add them to the Contact Directory
-View Numbers Dialed by Names from the Contact Directory
-Dynamic International Rate Table(Each customer can have his own price list using Brands)
-Rate Table in his Currency of choice
-Call Data Records including cost of each call and time based billing
-Call Data Records in his Currency of choice
-Switchboard (Displays live status of users phones and ongoing calls)
-Allows one click calling from GUI and direct to phone
-Call Parking sends calls to parking and then redirects to phone
-Allows transfers of calls
-Edit your Account setup
-Asterisk Billing and Management
-Edit voicemail setup including email and pin
-Create Time Based Dialing. Allows you to forward your calls based on time and day.
-Each user can have unlimited of SIP, IAX and Virtual Accounts
-Each user can have unlimited Prepaid Card Accounts linked to his userid
-Specify your hardware and change the viewable name of your accounts
-Temporary disable SIP, IAX or Virtual Account
-Manage your Incoming Public Numbers including Time based forwarding

User-friendly Administrator Web interface gives access to a range of functionality:

-Show Balance, Expenditure, Payments and number of Calls on each account
-Call Data Records including cost and Sales on each call
-Branding Module. Allows you to create Brands in any Currency
-For each Brand define Currency, Billing Increment, mark-up and connection charges
-Flexible Dynamic International Rate Table for Each Brand in any Currency
-Server Status (Displays live status of users phones and ongoing calls)
Show Peers.
-List of the last clients(SIP and IAX2) that have connected to the Asterisk server.
-Audit Trail. Show IP, Port and UserAgent for each call
-Manage your Incoming Public Numbers including Time based forwarding
Manage Trunks. You can use unlimited ZAP, IAX and SIP trunks.
-Time Based Trunk Dialing. Each trunk can have his own time based dialplan
-Temporary disable trunks
-Trunks can be rated after cost. Allow for cost based Dialing
-Define maximum concurrent outgoing calls on each trunk
-If lowest cost trunk is fully used (busy) the system will choose the next available trunk.
-Define unlimited outgoing routes and link them to your trunks
-Store cost of your outgoing route for each trunk for efficient cost control
-All outgoing routes are stored independent on the client price list
-Define customer price lists for each Brand and Currency
-Advanced customer management andportal management
-Integrated E-commerce module and web shop is available under GPL
-Define list of VOIP hardware commonly used
-Full Hardware Inventory. Store mac address and serial numbers of client hardware
-View and Store Customers payments
-Asterisk Billing and Management
-Manage Pre Paid and Post Paid customers. Full Credit control by User Account
-View important Server logs from web interface
-Define maximum concurrent calls on each Customer Account

Click Here to Download

Click Here to See System Requirements

Saturday, September 24, 2005

Pictures of Digium/Asterisk booth at VON Fall 2005

Kevin Fleming has posted a link on the "Asterisk User List" to pictures from the Digium/Asterisk booth at Fall VON conference. Thanks Kevin!

Click Here for Pictures

Announce: AstManProxy Version 1.12 Released

Hey folks --

I've released a new version 1.12, and there are a couple of developments to report.

I was contacted by Ron Arts with a patch to properly support the old/bad long-line and CLI output from Asterisk. I have applied his patch and tested it, and it seems to work well. Kudos to Ron for this important update!

Second, Nicolas Gudino, author of the Flash Operator Panel, contacted me about a couple of tweeks we needed to make to better support FOP. I took him up on his suggestions, combined with Ron's patch that better supports the CLI output, AstManProxy now works great with FOP! Nicolas is also adding support for talking to multiple servers using FOP/AstManProxy, so soon FOP can be used with a single instance of the proxy for controlling multiple Asterisk servers. That is some powerful stuff!

Last, I made a few other tweaks and changes. Please try out 1.12, especially with stuff that didn't work before, and report your results. I think this is a significant upgrade and should provide fully transparent compatibility.

Download v1.12 Here


Thursday, September 22, 2005

Announce: chan_capi-cm-0.6 released

"Hi all,

it took a while, but on I added the new release 0.6 of
chan_capi-cm driver.

Note: dial string and capi.conf has changed.

Version 0.6 Changes Are:

- added 'relaxdtmf'.
- more BSD compatibility
- correct PROGRESS handling
- added verbose text for capi info/reason error messages.
- use correct facility-selector for echo-cancel
- added application capicommand() for CAPI based applications
(removed standalone applications)
- capicommand(RETRIEVE) can now be called from other channels
- support ISDN hold (holdtype in capi.conf)
- added HOLD/RETRIEVE for Asterisk indications.
- added custom function VANITYNUMBER to convert letters into digits.
- added CAPI Line Interconnect (native bridging)
- use variable CONNECTEDNUMBER on Answer().
- set variable REDIRECTINGNUMBER on incomming call if it was diverted.
- removed obsolete thread mutex
- fixed dnid/exten/immediate handling on PtP.
- receive a fax via CAPI is now done with capicommand(receivefax|...) and added stationid...
- added config option 'immediate' to start pbx if no dnid has been received yet.
- support 'type of number' (numbering-plan).
- U-Law setting is now done in capi.conf instead of Makefile define.
- on hangup, use hangupcause from other channel or from var PRI_CAUSE.
- capi.conf structure changes: one own section for each interface, no global 'interfaces' any more. Section name will be interface name.
- dial string changed: parameters like 'b' not as part of number any more.
- send alert on alerting only (busy() and congestion() work now).
- better overlap sending (new parameter 'o' for dialstring to send only the first two digits with CONNECT_REQ only, the remaining digits and even digits following the dial() command, will be send as INFO_REQ/Overlap).

Have fun

Press Release: 5 Million Minutes Served: VoIP Provider Reflects on Successful Start of Operation

JAJAH VoIP provider growing quickly. High average revenue per user, more features to follow.

Vienna, September 19, 2005: Within three days of the initial introduction of JAJAH on July 1, more then 50.000 users have registered. 8 weeks later, an impressive 5 Million minutes have been served. 50% of the users rely on JAJAH for their free peer-to-peer phone calls, whereas 40% use JAJAH to call landline and mobile phones, 10% call other services (SIP, Skype etc.). JAJAH users can be found world-wide in 197 countries, with a remarkable ARPU (average revenue per user) of Eur 1.25, exceeding the ARPU of its competitors by an estimated 80%.

Built on service and philosophy

The impending release of the JAJAH full version (current beta version 2.2.08) will add more features to the already extensive list of services: e.g. free video messaging, the introduction of the Mac version plus inbound number for at least 10 countries. Existing features include text messaging, ringtones, skins and animations. This adds to the JAJAH mobile service that allows massive savings on international calls from your mobile phone. "We base our strategic philosophy on the beliefs of F. Jajah Watamba and understand our all-in-one communication tool as the practical application of free communication.", says JAJAH press manager Roman Scharf.

VoIP keeps on growing rapidly

On a more sober level, there seems to be no end to the world-wide VoIP hype: Following 72 billion minutes of VoIP use in 2005, the IDC expects an increase of up to 1.495 billion minutes until 2009 (2006: 183, 2007:367, 2008:760), with JAJAH undoubtedly taking a big bite out of this cake.

JAJAH technologies GmbH
Wohllebengasse 19
A-1040 Vienna/Austria
PR Manager
Roman Scharf

Announcement: HooDaHek 0.6 has been released.

Another fine post from the User list.

"So soon, you say? Well, the best laid plans of mice and men...

Steven BerkHolz is a pretty sharp stick and said to me, "Why don't you have HooDaHek change the CallerID when it looks up the name in the database on an incoming call?" Much head smacketh ensued, and as I made that change for Steven, I noticed that I had the way wrong version of hoodahek_dbhandle anyway.

Version 0.6 has the following changes:

- Got the correct version of hoodahek_dbhandle inserted, which has advanced error checking (yay) and also changes the CallerID in Asterisk if it performs a successful lookup in the HooDaHek database. Thanks to Steven BerkHolz for pointing out that rather obvious tidbit.

As always, information and download linkage available here:


Wednesday, September 21, 2005

SUCCESS - 512 Simultaneous Calls with Digital Recording

This is a nice little article I found on the user list. I love reading people pushing technology to higher limits and releasing anything they came across in there observations.

Full Post:

"List users,

Over the last few days we have been working with MCI's development lab to test our Asterisk setup. We were using a piece of hardware called an Abacus 5000 that is capable of creating and terminating thousands of SIP calls. Initially, we could not get past 64 simultaneous digitally recorded calls without having call quality issues including dropped calls. We identified an I/O bottleneck and rectified it by digitally recording to a RAM disk. Using this method, we were able to digitally
record 512 simultaneous SIP-to-SIP calls with 100% call completion.

Our plan is to use the MONITOR_EXEC hook to call a custom program that will copy files to the hard disk at call completion. This should be no problem when calling Monitor directly from the dialplan, but I need to know if there are any complications when digitally recording from app_queue or chan_agent. If anyone has experience with using MONITOR_EXEC and MONITOR_EXEC_ARGS with these applications, your input would be greatly appreciated. Digital recording seems to be the limiting factor when scaling an Asterisk system and we will share any advances we make with the community.

We also noticed some "Avoided initial deadlock" and "Avoided deadlock" warnings in the messages file that Asterisk generates. Our 128 and 256 simultaneous call tests each produced one of these warnings. Our 512 simultaneous call test produced 18 of them all within a 10 second span, each on a separate SIP channel. Can anyone shed light on what these warnings mean, how they can be avoided, and if they are something to worry about? They did not seem to affect the outcome of the testing in any way.

Details of our testing (hardware configuration, OS tweaks, system resource usage, reports from the Abacus 5000, etc.) will be made available over time. Please bear with me, as I am working on a deadline.

Thank you,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer"

Announcement: Brand New IPSwitchBoard

A new version of IPSwitchBoard has been released. This came through via the User list.

I am proud to announce that I have just released a brand new IPSwitchBoard totally rewritten from scratch. The new version has been built on the experiences gained by the previous versions.

Download for FREE:

IPSwitchBoard is totally customizable and will give you, among other things:

-Unattended/attended transfers.
-Park calls and retrieve/forward them again.
-Organize all your SIP, IAX extensions (automatically retrieved from Asterisk).
-Monitor all extensions, queues and Parked Calls.
-Dynamically log extensions in and out of queues.
-Set Do Not Disturb on Extensions and give a reason
-Set Call forwarding for extensions
-Set Dual call for extensions
-Monitor multiple calls on an Extension/Queue
-Monitor Extension online status
-Totally customizable
-Make you own skins with logo's etc.
-Make your own buttons - DND, Online, Queue Status, Call Forward, Dual Call, MWI etc.

IPDesigner is a unique tool for customizing and setting up IPSwitchBoard. With IPDesigner you can design your own IPSwitchBoard with Company logo and all the buttons you need for the Operator.

IPSwitchBoard works with .IPS files. These files contain everything needed for IPSwitchBoard such as bitmaps, server and extension configuration. You can build .IPS files with IPDesigner.When you have installed IPSwitchBoard you can start IPSwitchBoard by double clicking an .IPS file. If you right click an .IPS file you will have the options:

Open - will open the file in IPSwitchBoard
Edit - will open the file in IPDesigner
Configure - will start the configuration program where you can edit the server configuration.
Extract Bitmaps - will extract any bitmaps there's embedded in the .IPS file.

Tuesday, September 20, 2005

Announcement: asterisk-oh323: New versions 0.6.7 and 0.7.3

Hello all,

Updated versions of asterisk-oh323 are now available both for use with Asterisk v1-0 (version 0.6.7) and Asterisk HEAD/v1-2 (version 0.7.3).

Download from the usual location:


Monday, September 19, 2005

Announcement: HooDaHek 0.5 has been released


- Changed the format of the incoming call notification to be on one line, not two.
- Changed how the script sleeps between call notifications -- it now goes and outputs the line to everyone in the list, pauses, and then looks for a second line. Much better than before, where it was sleeping after every line.
- Changed the bot to use Unix::Syslog to log messages about its activities.

Information and download link here:



Sunday, September 18, 2005

Seizing the VoIP Opportunities, Rome 5 & 6 of October 2005

Jacob Fleming Group would like to invite you to the upcoming conference "Seizing the VoIP Opportunities". The event will take place in Rome, 5th and 6th October 2005.

This conference will bring you case studies on brand new VoIP pilot projects, service launches and visionary presentations from the industry's thought leaders like Telecom Italia, T-Mobile, City Telecom Hong Kong, Telenor and many others. It will assess, how telecom institutions can achieve the highest profits from streamlining their business model, explore new ways to upgrade infrastructure avoiding numerous risks and adding new revenue streams (besides cannibalizing existing ones) The five star setting and interactive format ensure that you will be able to discuss your most critical issues, gain inspiration and new ideas that you can apply directly in your business.

Your prestigious speaker panel includes:

Ni Quiaque Lai, Direcor of Corporate Development, City Telecom, Hong Kong Kerry Ritz, Managing Director, Vonage UK Alan Johnston, Distinguished Technical Member in MCI's Technology Strategy Group, MCI Massimo Coronaro, Technology Officer, Network Division, Telecom Italia Wireline Alan Duric, CTO, Telio Kennet Radne, Senior Vice President, Corporate Products & Services, TeliaSonera... and many other experts

Key topics:
- Strategic perspectives of VoIP and other IP services: Business models,'Triple play war'. How about quadruple and going wireless?
- Impact of VoIP on your networks and the overall technical infrastructure
- How secure is VoIP?
- Bringing new IP products & services to the market
- VoIP regulation: Will regulation play significant role in determining winners and losers?

Who should attend:
MoB's, CEO's, CTO's, CMO's, Vice Presidents, Heads and Senior Directors of Networks, Technology, R&D, Strategy, Marketing, Products & Services Development

Yours sincerely

Gabriel Meixner
Marketing Manager
Jacob Fleming Group - Europe

Tel.: +421 2 5825 2731
Fax: + 421 2 5825 2778

Tuesday, September 13, 2005

New York Asterisk User Group - Established

Efforts are underway to setup a New York Asterisk User Group which is free, and open to the public.
We're currently looking for members who share a passion for creating, and discussing, interactive telecommunication services with Asterisk and other Open Source technologies.

* Meeting Dates: TBD
* Site secured: Columbia University NY, NY
* Visit for more information:
* Subscribe to the mailing list:

Skype purchased by Ebay 2.6 Billion

Here is an excerpt from a news article. As expected a handful of analysts are questioning the purchase. I wonder if they will integrate skype into there site to have a more interactive customer service option for there sellers?

"Sweden Some analysts are questioning the price eBay is willing to pay to buy Internet telephone company Skype (SKYP) Technologies.

San Jose-based eBay announced this morning that it would pay at least two-point-six (b) billion dollars in cash and stock for Skype.

But eBay says the total value of the deal may climb to as much as four-point-one (b) billion dollars depending on whether Skype meets a series of performance targets over the next three years."

Full Article Here

Announcement: ruby-agi 0.0.2 released


I have released Ruby Asterisk Gateway Interface (ruby-agi) v0.0.2b. Any feedback, bug report, suggession, feature request is most welcome.

ruby-agi homepage:

Download ruby-agi v0.0.2b here:

Mohammad Khan

Friday, September 09, 2005

Asterisk Management Portal (AMP) 1.10.009 released!

Hello all,

Asterisk Management Portal 1.10.009 has now been released. This exciting new version has several notable additions (listed below).

The AMP homepage is Here you'll find links to the download, install guide, and documentation wiki.

As usual, please use amportal-users mailing list for discussions about AMP:

AMP 1.10.009 changes:

- Optional separation of Devices and Users. Devices are endpoints (ie: phones), and can be Fixed (to a user), or Adhoc. Users are extensions, with options like voicemail. A user can log in to one or more Adhoc devices by dialing *11, and log-off by dialing *12.

- "Custom" device technology support - this means devices that are not configured directly in AMP's admin can still be used (ie: SCCP)

- Asterisk Recording Interface (ARI). ARI is a php interface to Voicemail and Monitor recordings. (written by

- RingGroups now use strategies: Ring All (default), Hunt, Memory Hunt

- DID Routes re-written as Inbound Routing. This allows for DID specific fax emails and call answering options.

- Queues can now play a "welcome" message to callers upon joining.

- HINT priorities for FIXED devices

- Interface translated to French, German, Italian, Spanish

- FOP .21

- FOP button layout can now be sorted by last name or extension number

Ryan Courtnage
Director & CTO
Coalescent Systems Inc.

Announcement: Flash Operator Panel 0.23 released

Dear all,

I'm happy to announce the Flash Operator Panel 0.23 release. FOP is a switchboard type application for the Asterisk PBX. It runs on a web browser with the flash plugin. It is able to display information about your Asterisk box in real time. It is included in AMP, Asterisk@home, etc. You can grab the latest version from

The incomplete list of new features is:

* Internationalization support (thanks to everyone who contributed a language file! If your language is missing, please contribute with the translation. It is a small file, a couple of minutes worth of your time)

* Command line options. You can specify the logdir, pidfile, debug level and much more from the command line.

* The web_hostname parameter is now optional. It eases the installation a lot! All systems that include FOP installation and configuration scripts can now leave the field commented, and the client-server connection will just work(tm). No need to fiddle with ip addresses, hostnames, etc.

* Popups via UserEvent can be restricted to one button/viewer only.

* Added font and shadow color parameters for button labels, text legends, clid and timer.

* Added event_mask parameter to filter unwanted events from the manager

* Improved debian init script. Thanks to Tzafir Cohen.

* It uses a lot less CPU than previous versions on heavy asterisk boxes

* Improved support for parking when using native sip transfers

* Minor bugfixes

* A stupid buglet that I don't know yet about, that will force me to release 0.24 in a short while.

The upgrade instructions are on the tarball UPGRADE file. Remember to upgrade the .swf file and to flush your browser cache!

Many thanks to everyone who provided feedback, patches, ideas and suggestions.

I'm also saving money to attend to Astricon Fall. Please consider a small donation to help me cover the travel expenses. (my deepest thanks to everyone who already donated to the project!)

PS: Just to let you know, I'm playing now with new tools to develop flash clients: swfmill and mtasc. They are great tools, it will ease and make the development faster. But the port is not easy, I will work slowly on that. If you are a flash OOP actionscript fan and want to help, please let me know.

Nicolás Gudino
Buenos Aires - Argentina

Thursday, September 08, 2005

MINNESOTA: TwinCities Asterisk Users Group -Saturday 9/10/2005

Good Evening,

The next Asterisk Users Group meeting has been scheduled for September 10th at 11:30am, this comming Saturday.

Meetings are held monthly on the second Saturday of each month, excluding July and December.

Sound Choice Communcations has moved to Bloomington Minnesota, just 1/2 mile west of the Mall of America. The New address is: 7839 12th Ave S, Bloomington Minnesota 55425. We are just South of Hwy494 on 12th Ave. -12th Aveune is one exit West of Hwy 77 (Ceder Ave).

Meetings are held at Sound Choice Communications LLC...

We are always looking for help with meeting topics. If you feel like taking the lead, please do and simply let me know if you need anything.

Meeting starts at 11:30am and parking is available in the rear of the building. Runs about 2 hours or less, and we'll order Pizza to the meeting for lunch.

Look forward to seeing you there.

PS: Time permitting, Sound Choice will be selling tons of its extra inventory. Extra monitors, spare computer equipment, VoIP equipment. We've just moved these items and realize that moving our stuff is too much work.

Announcement: ASTPP-1.2-Beta Released

Good Day,

I'm please to announce ASTPP (Asterisk Post Paid) billing version 1.2 Beta. I've spent the past months working on this software and have added several new features. Here is a partial list of new features:

-Partially redesigned web interface
-Automatic DID mapping
-Least Cost Routing
-Improved support for tying into webstores

Screenshots available @

As always it is licensed under the GPL. For more information check the wiki @

We plan on bringing this to a release as soon as possible but the beta has proven stable and is currently being used.

AstriCon Update: Please Register ASAP - Free Phones

_Book Your Hotel Room Today_

We're now a little more than a month away from AstriCon 2005 - The Asterisk Conference and Exhibition. We need everyone who plans on attending to register with the Hyatt ASAP to ensure we have enough hotel rooms. (Last year in Atlanta we over-booked the hotel by over 150 people.) If you have already registered for the conference, you can book your rooms by going to:

Click the link that says "Make A Hotel Reservation At The Special $114.00/night Rate".

-Free IAX Phone To The Next 50 Paid Conference Registrations-

Ipsando, creator of the IAX Phone, has agreed to provide the next 50 paid conference attendees* with a free licensed copy of IAX Phone Pro _plus_ a free VoIP handset. Winners will be notified by email and can pick up their free CD and handset at the Ipsando booth. For a beta version of the phone and a list of features, please see:

_Any More Speaking Proposals?_

If you would like to speak at AstriCon 2005 and have not yet sent in a propsal, please do so ASAP. We need your proposal, biography, and a head-shot photo (see the instructions on the AstriCon 2005 site: Speak At AstriCon). We will begin making final selections this week.

_Any Questions?_

If you have any questions about AstriCon 2005, please let us know. Email We want to make this the best AstriCon yet.



Tuesday, September 06, 2005

IP PBX shares over 91 percent of shipments by 2009

I received an email this morning containing a link to this story. I talks about the growth of IP PBX. I hope this keeps on track, from this we know that a certain portion of these will be Asterisk :)


"United States - While the total PBX market is projected to grow by 6.6 percent CAGR through 2009, the traditional PBX will decline rapidly and the IP PBX will continue to soar throughout the period. By 2009, server-based IP PBX shipments will grow from 9.5 million lines to 28.1 million, accounting for over 91 percent of total PBX shipments. In-Stat reports that this year alone, shipments of IP lines are expected to exceed those of the traditional PBX."

1st beta of ruby-agi is out

I just released first beta version of ruby-agi (ruby-agi-0.0.1). ruby-agi is an Asterisk Gateway Interface (AGI) written in Ruby.

Project homepage: you can download ruby-agi-0.0.1 from:

Feel free to try it.
Your feedback, bug report, feature request, suggession would be greatly appreciated.

Mohammad Khan