Asterisk VoIP News

Monday, October 31, 2005

Announce: Asterisk 1.2.0-beta2 Released!

The second beta of Asterisk 1.2.0 has been released! It is available from the FTP servers, as well as the Digium CVS servers (under the 'v1-2-0-beta2' tag).

This beta includes a large number of improvements over beta1, including:

* Many, many bug fixes
* Documentation and sample configuration updates
* Vastly improved presence/subscription support in the SIP channel driver
* A new (experimental) mISDN channel driver
* A new monitoring application (MixMonitor)
* More portability fixes for non-Linux platforms
* New dialplan functions replacing old applications
* Significant deadlock and performance upgrades for the Manager interface
* An upgrade to the 'new' dialplan expression parser for all users
* New Zaptel echo cancellers with improved performance
* Support for the latest OSP toolkit from TransNexus
* Support user-controlled volume adjustment in MeetMe application
* More dialplan applications now return status variables instead of priority jumping
* Much more powerful ENUM support in the dialplan
* SIP domain support for authentication and virtual hosting
* Many PRI protocol updates and fixes, including more complete Q.SIG support
* New applications: Pickup() and Page()
* ... and many more I'm sure I've missed!

We ask all interested community members to download and install the beta release (on a non-production server) and report their findings via our bug tracker. Please be sure to read the UPGRADE.txt file in the distribution before upgrading your server, as there are a large number of changes that you will need to be aware of (some of them are not backwards compatible with the 1.0.x releases).

We want to extend our thanks to all the community members whose contributions have made this release possible; without their support, testing and other involvement we would not have reached this milestone so soon!

Announce: HooDaHek 0.7 Released

Version 0.7 of HooDaHek, the CallerID Notification System for Asterisk, has been released. This is the last version with this database schema; the next will be breaking things into many more fields of information, including a lot of custom actions for calls (sending calls to VM by CLID, ignoring calls, etc.)

Also announcing the new HooDaHek website for the files, docs, and more information:

Details about Version 0.7:


- Added in Jabber support.
- Added in SMS support for US Cellular, Midwest Wireless, Verizon, and AT&T phones.
- Added in email support. You can now have notifications emailed anywhere. Handy for EmailToSMS gateways and/or pager alerts.
- Added a command, "last", to HooDaHekBot's vocabulary. The "last" command, when typed with no parameters, will show you the last incoming phone call. When typed like, "last:X", where "X" is a number, it'll show you the last X calls. IE: "last:10" will show you the last 10 calls.
- Fixed a bug with the Month being off on the call notification.
- Added signal handling for proper cleanup on exit.
- Added in massive debugging and logging for AIM connections as well as more error checking for disconnects, etc.
- Changed the sorting of the results of the "name:" and "phone:" commands to be sorted by alpha and number.
- Redid the entire configuration section of because it was long and incomprehensible.

- Switched from interfacing with the 'mail' command to using SMTP to notify of a new addition to the database.
- Formatted the phone number now in notifications. Changed the wording and subject slightly.

- Changed the GRANT statement in the SQL file so it wouldn't grant permissions on anything more than the CDR database.
- Decided to release HooDaHek under the "MIT License", which allows for free redistribution and modifications, for any purpose. Knock yourself out.



Release: Signate Adds T1 Interface to Telephony Server 5000

SAN FRANCISCO, CA/TORONTO, ON, October 31, 2005 - Signate, the leading provider of VoIP telephony solutions that combine high performance hardware and open source software, and Sangoma Technologies Corporation a leading provider of connectivity hardware and software products for VoIP, TDM voice, WANs and Internet infrastructure, today announced that Signate has certified Sangoma's A104 AFT T1/E/J1 card for use in Signate's Telephony Server 5000.

"A single module of Signate's Telephony Server 5000 easily handles 276 calls from a dozen T1s with three of our A104 AFT T1/E/J1 voice/data cards," said David Mandelstam, president and Chief Executive Officer of Sangoma Technologies. "That leaves plenty of horsepower for SIP calls and other Class 5 services."

"Our engineering capabilities with this card and others to shortly follow raises the bar for our competitors, and offers a new price/performance standard unparalleled in our industry," adds Mandelstam. "We continue to take major steps into the Open Source VoIP arena, bringing new products to market and setting solid examples others are now following. Having Signate certify our hardware demonstrates real industry trust."

Signate's Telephony Server 5000 is the industry's first high performance PBX or softswitch built for the open source PBX environment, rated at 2500 simultaneous SIP calls per module. Sangoma's A104 provides full speed 132 Mbps PCI bus transfer with hardware echo cancellation to unload the host CPU in demanding PBX/IVR voice applications.

"We chose Sangoma's product because they are best in class, with enterprise grade cards that are engineered to share interrupts properly between themselves and other PCI compatible devices, so we can use multiple cards in each chassis," said William Boehlke, Signate's Chief Executive Officer

Sunday, October 30, 2005

Announce: Asterisk-Java 0.2-rc2 released

Asterisk-Java 0.2-rc2, a Java control for the Asterisk PBX, has been released.

The Asterisk-Java package consists of a set of Java classes that allow you to easily build Java applications that interact with an Asterisk PBX Server. Asterisk-Java supports both interfaces that Asterisk provides for this scenario: The FastAGI protocol and the Manager API.

The 0.2-rc2 release candidate focuses on the new features of the Asterisk 1.2 series though it still supports Asterisk 1.0.x.

The changes include:
* Bug fix for variables in OriginateAction (AJ-15)
* Support for FaxReceived event from spandsp (AJ-20)
* Better integration with Spring Framework via SimpleMappingStrategy and AGIServerThread

Asterisk-Java is used in several commercial environments and by the following Open Source projects:
* Asterisk-IM
A plugin for the Jive Messenger XMPP (jabber) server. It provides integrated presence between your IM client and phone, notification of incoming calls by IM and originate calls from supported IM clients.
* Asterisk Desktop Manager (ADM)A desktop application that will allow for automatic on-call volume reduction, one click dial from clipboard, integrated phonebook and more.

Asterisk-Java is available under Apache 2.0 license at

Thursday, October 27, 2005

Release: SANGOMA Technologies "A104d" Provides Carrier Grade Audio To The Soft PBX Industry

Unique engineering design uses a common PCB to host hardware-based echo cancellation and voice enhancement for all voice platforms

LOS ANGELES - October 27, 2005 - Sangoma Technologies Corporation (TSXV: STC) (, a leading provider of connectivity hardware and software products for VoIP, TDM voice, WANs and Internet infrastructure, has launched its new A104d Series of cards that provides hardware-based carrier grade echo cancellation and voice enhancement.

The A104d includes a miniature voice enhancement sandwich board. The voice enhancement capabilities added to Sangoma's standard AFT-based A104 card include: G.168-2002 echo cancellation with 1024 tap/128ms tail per channel on all channel densities, DMF encoding/decoding and tone recognition, voice quality enhancement and adaptive noise reduction. Other industry-leading features will be added in the future.

"Designed at Sangoma's research and development labs, the PCI card is engineered for today's demanding soft PBX, IVR and VoIP applications, such as Asterisk, Yate, and OPAL, offering a new price/performance standard unparalleled in our industry," says Sangoma Technologies President and CEO David Mandelstam. "Over the next few months, the same voice enhancement functions will become available on all our voice handling products, from individual single line telephone connections up to T3/E3."

Distinctive A104d features include:

* Support for 1024 taps (128ms) of echo tail handling on each and every channel (DS0) meaning that troublesome delayed echoes are properly handled.

* Dynamic and selective activation of echo cancellation, making the system ideal for mixed voice/data applications.

* Octasic Semiconductor's internationally deployed carrier-grade echo cancellation solutions, delivering unprecedented voice quality. With Octasic's advanced voice enhancement features, users enjoy the highest standard of quality on all calls.

* The same PCI interface, architecture and digital path as Sangoma's T1/E1 cards, meaning no motherboard or compatibility issues and ultra-reliable interrupt handling.

Visit Sangoma Technologies at the Internet Telephony Conference & Expo at Booth # 522

Click Here for More Information

Wednesday, October 26, 2005

News: New Bug Marshal

Note: Kevin Fleming has announced another new bug marshal. His post is below:

Alex Lopez (username 'opsys') has joined our bug marshal team and will be helping to keep bugs moving through the system and get them tested/confirmed, so please welcome him and give him lots of work to do!

Announce: A2Billing - AreskiCC v3 New Release

Dear Friends,

Great day for the callingcard-lovers !!!
I am pleased to release the version 3.0 of AreskiCC !!!

Little unexpected change, we got a new name... bit more serious "A2Billing" Many many features have been added, lot bugs solved and a bunch of good enhancements made!

The key newest features :
- Full MYSQL support
# calls compare
# monthly traffic
# daily load
- FASTER RATECARD CREATION (range, interval)
- Register_global = Off :D
- Recurring service : Apply batch process of certain card.
- CONFIGURATION /etc/asterisk/a2billing.conf
- SUPPORT SEVERAL CONFIGURATION DeadAGI(a2billing.php|%idconfig%)
- and more...

Other good thing, we have an handbook :D It's covering mainly installation for the moment but I will complete the part for the user guideline pretty soon.
Any help (documenting, dev...) would be greatly appreciated, so if someone is willing
to help, please contact me !!!

I am sure you will enjoy this new version!
Have fun and don't forget to send me some feedback,

Announce: fax2mail script update (includes hoodaheckcompatibility)

Thanks for the great feedback! We now have an updated fax2mail (version 2.0) for download (at Fixes include:

1. Improved detection of the number of fax pages
2. Handling of "!" character in the name (for those users of the "hoodaheck" module)
3. Correction of several syntax errors in the bash script.

For those of you new to faxing with asterisk, fax2mail is a script which is called by spandsp upon arrival of a fax. It will convert the image to PDF/EPS/TIF format, and email the fax to a user (just like voicemail), include numerous fax information, etc.


Announce: Digium Announces Wildcard TDM2400P (24 Port Card)

Note: Here is the Press Release from the Digium site.

Digium Inc., the creator of Asterisk and pioneer of open source telephony, launched the Digium Wildcard TDM2400P, the most dense and scalable card available for building an Asterisk-based telephony system for SOHO and SME environments. The 32-bit 33MHz PCI 2.2-compliant card with Digium's patent pending VoiceBus architecture, supports quad-FXS station and quad-FXO office interfaces for connecting analog telephones and lines through a PC, without taking up numerous PCI slots.

"The Digium 24-port card offers the highest analog density available in a PCI card and can scale up to 48-ports with two cards and two slots," said Mark Spencer, president of Digium. "With its flexible scalability features, our 24-port card is the best hardware card available for small and medium businesses looking to build an inexpensive, sophisticated VoIP telephony solution without compromising the use of multiple PCs."

The Wildcard TDM2400P replaces the requirement for a separate channel bank and T1 interface cards while offering superior echo cancellation on both FXO and FXS interfaces. The quad-FXO and quad-FXS modules are interchangeable allowing the combination of interfaces up to six slots for 4-port FXS or FXO modules. With this new card, small and medium businesses can benefit from features such as high density in fewer PCI slots and an industry standard 50-pin Amphenol connector for easy installation.

Support and Availability
The Digium Wildcard TDM2400P will be available from Asterisk resellers and distributors worldwide beginning December 2005. For more information, please contact or call 1-877-LINUX-ME. All Digium products are backed with a two-year limited warranty, including installation and troubleshooting support. Users can also purchase an optional one-year extended warranty.

About Digium
Digium is the creator and primary developer of Asterisk, the industry's first Open Source PBX and Asterisk Business Edition, the professional-grade version of Asterisk. Used in combination with Digium's PCI telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over IP, TDM, switched and Ethernet architectures.

Digium solutions reduce the costs of traditional TDM and VoIP implementations through open source, standards-based software and innovative hardware solutions, including legacy PBX, IVR, auto attendant, and next generation gateways, media servers, and application servers. Digium hardware supports traditional voice and data protocols, and packet protocols such as IAX(Inter-Asterisk eXchange), and SIP VoIP.

Digium provides a highly refined selection of quality hardware and software products, developed and implemented using innovative engineering techniques (primarily open source development). A full range of professional services complement these product lines, including consulting, technical support, and customer software development services. The open source communications revolution is here, and Digium is leading the way.

Click Here for more Details

Astricon: Materials Online

Ollie posted where to find some of the latest Astricon materials for anyone who missed it.

Registered attendees will get information about the material soon.
No videos where recorded this year.

The 1.2 presentation I made together with Kevin has been available for a while at and will be updated soon.


News: Groups challenge FCC's VOIP wiretapping rules

A group of privacy advocates and technology companies on Tuesday filed court papers to challenge a ruling by the U.S. Federal Communications Commission (FCC), saying it overstepped its authority by requiring VOIP (voice over Internet Protocol) providers to allow wiretapping by law enforcement agencies.

The groups, including advocacy groups the Center for Democracy and Technology (CDT) and the Electronic Frontier Foundation (EFF), argued that an FCC's ruling on VOIP could introduce security vulnerabilities into VOIP services, could drive up costs for customers, and could open up additional Internet applications, such as instant messaging, to wiretap rules.

The August 2004 FCC ruling requires VOIP providers, by early 2007, to build in technology that complies with a 1994 telephone wiretapping law called the Communications Assistance for Law Enforcement Act (CALEA). But adding such functionality to VOIP could introduce security holes by increasing the complexity of the code, and it could open up vulnerabilities to sophisticated hackers, said Susan Landau, a distinguished engineer at Sun Microsystems Inc.

Click Here to Read the Full Story

Announce: New Zealand Asterisk Users Group

Hadley Rich has posted about a Asterisk group that is now active in New Zealand.


Since we're doing this...

There is now a New Zealand Asterisk Users Group set up.

There is a wiki and mailing list at both are sparse at the moment and could do with some input.

If you're in New Zealand (or not) and interested in Asterisk then sign up and get contributing!

Thanks, and please excuse the spam.


"I can't decide whether to commit suicide or go bowling."

-- Florence Henderson

Monday, October 24, 2005

News: Aruba unveils portable access point for VoIP

Trixter from the User List has post this nice story about a VoIP access point that looks to be very mobile.

Aruba Networks will announce this week an enterprise-class mobile access point that will put users at the edge of their corporate network no matter where they are located.

When it becomes available, the hardware-software solution will set up a tunnel across the Internet to the Aruba Centralized Mobility Controller sitting behind a corporate firewall. It will also include a choice of two access points -- a portable model, AP-65, measuring three-by-three inches; and model AP-41, designed for home office.

"It is as if the enterprise wireless went with them. It pops up as a corporate hot spot wherever you plug into," said the Aruba Networks founder Keerti Melkote.

The solution would look the same to end-users as a VPN for data. However, the solution is more manageable than a VPN because it includes an access point, according to Craig Mathias, principal with the Farpoint Group.

Click Here for Full Story

Sunday, October 23, 2005

Announce: Asterisk .NET Released!

A port of Asterisk-Java to C# for Microsoft's .NET Framework 2.0 (Visual Studio 2005 RC) and anything else that implements the basic portions of the framework required to use this.

The Asterisk .NET library consists of a set of classes that allow you to easily build applications that interact with an Asterisk PBX Server.

Asterisk .NET supports both interfaces that Asterisk provides for this scenario: The FastAGI protocol and the Manager API.

Asterisk .NET is compatible with Asterisk 1.0 and 1.2.
The FastAGI implementation supports all commands currently available from Asterisk.

The Manager API implementation supports receiving events from the Asterisk server (e.g. call progess, registered peers, channel state) and sending actions to Asterisk (e.g. originate call, agent login/logoff, start/stop voice recording).

Click Here For More Information

Click Here to Download Latest Version

Thursday, October 20, 2005

Open Source Asterisk Management Portal Project Celebrates One Year Anniversary

Note: I found this release about AMP(Asterisk Management Portal) on a newswire dated on the 12th of Oct.

"A year ago today AMP was unleashed to the SourceForge(R) code repository website ( The response was immediate and favorable. Interest in AMP rose steadily and it quickly became the de facto freely available Asterisk GUI. The inclusion of AMP in the immensely popular Asterisk@Home distribution contributed to this success, as did the countless contributions of the AMP developer community and user base. The AMP developer community grew from 2 to the 12 it numbers today and the amportal-users mailing list has over 400 subscribers.

AMP's feature set has grown from that of a modestly equipped IP PBX to a full-featured IP PBX that rivals many proprietary offerings that are Asterisk-based. Combined downloads of AMP and Asterisk@Home are fast approaching the quarter-million mark. In the coming year the push is underway to release version 2 of AMP. According to Ryan Courtnage, Director & CTO of Coalescent Systems, AMP 2 will adopt a more modular design. Courtnage also shared that "Work is underway to reduce the number of dependencies thereby making installation of AMP easier. And with the release of Asterisk 1.2, there will be lots of new goodies in store".

On this one year anniversary of AMP, Coalescent Systems would like to assure the AMP user base and developer community that the development process will continue to be open and value justified. It is our sincere hope that continued interest will maintain AMP's reputation as a robust and innovative next generation phone system."

Click Here for Full Press Release

Update: Asterisk Community Participant | Katrina Refugee UPDATE

Note: Here is an update from JR Richardson. He is a Katrina survivor and I wanted to post his update for people that are following his family's story. Also I would like to add that it is very refreshing to hear what Digium did in his wake. Great work!

Hi all,

Thank you all for your replies of hope, and advice for recovering flooded computer equipment. I was not able to recover ANY electronic components. There was 5 foot of water sitting in my home for over a week. The water was laden with very corrosive contaminants and heavy sludge. Literally this water ate the conformal coating off a lot of ckt boards then heavily corroded and oxidized any metal and solder. I struggled to recover data from hard drives but did manage to get a couple to work after cleaning.

Silver Lining:

I lost a few thousand dollars just in Digium hardware alone, so I contacted Digium and let them know that I was out of commission for a while till I could get another lab setup. In my most humble manner I requested card replacement or discount on a few items just so I could get back up and running, to my surprise, Digium would not accept any money from me and sent several cards and components free. They thanked me for my participation and expressed compassion for my loss. Digium is a class act and have helped me rebuild some of what Katrina knocked down. I will forever be grateful and never forget Digium's Support in my time of need.

Thank you; Mark and Malcolm, you guys are the best.

I have already relocated my family to Lafayette, LA and we are living in the home we intend to buy soon, just need to run some extra power ckts to the spare room and setup a new lab. We took in a family that lived across the street from us in St Bernard and they are living in my future computer lab, so it will still be a bit before I get up and running but will soon.

JR Richardson

Wednesday, October 19, 2005

Announce: New ISDN architecture available for Asterisk

Matteo Brancaleon has posted on the List with details of a new ISDN protocol.


A new ISDN architecture, called vISDN, has been developed to fully support EuroISDN protocol with HFC based cards: HFC-S PCI, HFC-4S and HFC-8S (with HFC-E1 and HFC-S USB support coming soon).

vISDN is not based on Zaptel, libpri, chan_zap, zaphfc, qozap, etc... but has been designed from scratch to be a standard compliant EuroISDN implementation plus a channel crossconnector, plus protocol analysis support thru Ethereal, plus a ppp terminator, plus other stuff :)

Main Features:
- Open, modular, flexible and versatile architecture
- Fully GPLed
- Full support for PRI and BRI
- Full support for Network and Terminal Equipment role
- Traffic analysis with (patched) Ethereal
- E-channel sniffing
- D-channel sharing between applications (in TE-multipoint mode)
- Good integration with latest 2.6 kernels and extensive usage of their newer features (e.g.: sysfs)
- Protocol stacks are implemented following the finite-state-machine models described in the ETSI specification for better compliance/debuggability.
- Termination of PPP connections without exiting from the kernel
- Takes advantage from HFC's framer for HDLC traffic
- Preliminary/experimental support for hardware PCM bus
- Preliminary support for hardware channel bridging
- Support for dynamic (optionally automatic) activation/deactivation of layer2 (DLC connection)
- Unintrusive with respect to the Linux kernel and Asterisk (no patches needed)

Missing Features:
- Echo cancellation (likely going in in the next release)
- Explicit Call Transfer

Please note: This is a Linux-only, 2.6-only architecture which supports only EuroISDN. It is currently in beta-stage, brave testers will be more than welcome; people who want to contribute will be welcome too. You will not need to sign a disclaimer to participate but you will receive good and valuable consideration, plus discrete amounts of beer :^)

vISDN will undergo a layer2/layer3 certification by an independent lab and will have a "Declaration of conformance" valid in the EU territory. AFAIK, the declaration is valid for the product as a whole, so, it is to be seen if the declaration could be extended to "mixed" products.

Please see for further information. If you are near Milan Italy, vISDN is live at SMAU expo. We will be happy to show the driver working live and exchange comments and ideas about it(Voismart - hall 12 booth H22)


News: Is P2P SIP Poised to Out-Hype Skype?

Note: I found this over at the Voxilla about P2P SIP, interesting read. I have opened this topic up for discussion. Please comment about your thoughts on this.


"While Luxembourg-based Skype's closed-source software currently holds a huge lead in peer-to-peer VoIP, enthusiastic backers of the open source SIP protocol believe that an under-development peer-to-peer SIP approach is poised to catch up with Skype and even surpass it.

"SIP is an open standard and that has a longer term impact than a closed network philosophy," says Eric Lagerway, co-founder of the softphone company Xten (a company he left earlier this year) and founder and CEO of Vocalscape. "While a closed network can foster innovation it does nothing for the long-term for IP communications as a whole. In the long term you're better off with an open standard and that's what peer-to-peer SIP is."

By relying on an open source development model similar to that used with the popular Asterisk PBX, Lagerway believes, the sheer amount of engineering and software talent applied to it is significantly greater than the effort going into furhter enhancements of Skype.

"Taking Skype past where it is today will be hard," Lagerway says, "because there are only so many people working on it while all of the working groups of the IETF (Internet Engineering Task Force) are working on [some aspect of] SIP."

The benefit of peer-to-peer SIP is simplicity. It eliminates the server from the network by distributing the SIP methodology throughout the network itself.

Click Here for Full Article

Astricon Report Update: Xorcom's Two New Products

Note: Here is some updated information about the two products released by Xorcom. At the time there was not information about the new products posted to there website. Here is the updated info from there site:

"At Astricon, the only conference dedicated to the Asterisk Open Source PBX, Xorcom announced the Astribank-8, the first channel bank designed specifically for Asterisk, and the TS-1, a complete Asterisk server "out-of-the-box"."

Click Here for More Information

News: World's smallest VoIP PBX?

I got this email with a link to an article on Linux Devices. They seem to have Asterisk running on an embedded Linux distribution.


"A project to create an embedded Linux distribution around Asterisk, an open source PBX (private branch exchange) software package, is demonstrating a VoIP PBX system running on a tiny Gumstix SBC (single board computer) at the AstriCon trade show this week in Anaheim, Calif. The AstLinux project's demo may be the smallest PBX system ever created.

The AstLinux project was founded about a year ago by Kristian Kielhofner, a 21-year old hacker who runs KrisCompanies, a consulting firm based in Lake Geneva, Wis. The project aims to create a minimal embedded Linux distribution that makes it easy to install and experiment with Asterisk.

Click Here to Read the Full Article

Tuesday, October 18, 2005

Announce: Free DIDs on

Note: Matthew Simpson posted on the User list about free DIDs being given away.

GoIAX, the Asterisk community's free IAX provider, is offering free US DIDs now. I loaded about 175 dids in and put up a very beta sign in page.

Unfortunately I had to restrict the free us/canada outbound calling back down to toll-free only. There was a lot of war dialing and prank calling going on. I'm working on some stuff to hopefully curb that kind of stuff down so I can unrestrict outdial again, but this is the problem with free.. there is always someone that will abuse it.

If anybody has any ideas on how to keep the abuse down let me know. The best ideas I have now is to only allow a certain amount of calling per month, add velocity checking, and somehow put some accountability into the sign up process to keep the prank callers and multiple account abusers away.

Matthew Simpson
GoIAX --
TxLink --

Monday, October 17, 2005

Announce: Ruby module for the Asterisk Manager Interface

Chris has posted a new module for the Asterisk Manager Interface.

I have just released the first version of Rami, a ruby module for the Asterisk Manager Interface. It includes a client library and proxy server for sending multiple simultaneous requests with just one open connection to asterisk.

One of the unique features is that the proxy server stores internal events into queues which can be retrieved or searched by value. For example with the Originate command, if you use it with Async, it will return immediately and the proxy server will store the associated events in the queue which can be queried at a later time. WIthout Async the Originate command will block until it is finished, returning all the events at once.

Rami is distributed as a Ruby Gem.

You can download it and view the documentation at:


Announce: AstBill- Released

The AstBill project has released version of its open-source billing and VOIP management platform for Asterisk. There are many new features in this release. AstBill is also a maintenance release that fixes problems reported using the forums and the bug tracking system.

We STRONGLY recommend you to update to the latest version as AstBill. The software is under a very fast development schedule mainly thanks to feedback from the fast growing user community.

AstBill is a Web Based Billing, Routing and Management Software for Asterisk and VOIP based on Drupal and MySQL. AstBill Provides pre and post Paid VOIP Billing Services. The aim of AstBill is to completely automate Asterisk, call management and VOIP billing from start to finish.

Key benefits is Open Source, Credit Control on outgoing calls, ease of use and the User Management and call routing module. AstBill is fully themeable and skinable.

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

There is also an AstBill Live CD available. This allows you to run Asterisk and AstBill from your CD drive. No installation needed.

Some of the new and improved features of AstBill-

-Implemented full support for H323
-Improved web interface for accounts management. You can now choose between DISABLED, REALTIME, STATIC and ANI/CLI authentication.
-Improvements in web interface
-Fixed problems reported using the forums and the bug tracking system.
-Improved Debug output on Perl agi scripts
-Minor Update to MySQL database schema
-Updated extensions.conf added example used when Asterisk and AstBill integrates with SER
-Implemented stronger caller authentication security
-Improved Multi Tenant functionality
-Rate Table in 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 user's phones and ongoing calls)
-Allows one click calling from GUI and direct to phone

Click Here to Download

Friday, October 14, 2005

Astricon Report: Mark Spencer's Keynote Address

Another great read from Mr. GP. The following is a summary of Mark's keynote address at Astricon 2005:

"Mark Spencer, the founder of Digium and the Asterisk Open Source PBX project gave the Keynote address this morning to a crowd of over 500 people gathered at Astricon 2005 in Anaheim California. Mark's talk focused on the key differences between Asterisk as an Open Source project and closed source solutions. He went on the talk about briefly about the upcoming release of Asterisk version 1.2. But an announcement of the release, which was anticipated by some attendees was not made. Mark and other speakers at the conference indicated that new features recently added still needed to be fully tested, and bugs were still being addressed.

Despite the lack of a new release, the conference has generated a great deal of excitement from attendees. The Astricon conference gives the Asterisk developers and user community a chance to meet, exchange ideas and have fun. Digium hosted an after conference party on Thursday night at which users and developers from around the world drank free beer and talked about Asterisk late into the evening.

Next years conference has been designated to be held in Dallas TX. Dates for this event should be released on the website by Nov. 15.

Astricon Report: Xorcom Astribank 8 - USB Channel Bank for Asterisk

Another report has come in from Gerald P. This time he is looking at a Channel Bank from Xorcom.

"A company called Xorcom is displaying an interesting FXS channel bank at this year's Astricon in California.
The Astribank 8 is an 8-port FXS channel bank that connects to an Asterisk server via a USB interface.
With an MSRP under $500, this device is seen by Asterisk as 8 Zapata (ZAP) channels. The device also features 2 relay outputs and 4 contact inputs which are accessible via the Asterisk Dial Plan. This feature is interesting in a number of creative applications including actuating door strikes and interfacing with alarm systems via an Asterisk phone system.

The company's website ( does not yet have any additional information regarding this product or its availability, but it looks interesting. If they were to offer and FXO version, that would start to be really interesting.

Gerald P.
VoiceIP Solutions

Thursday, October 13, 2005

Astricon Report: Intriguing new product at Astricon

Gerald Pickford of VoiceIP Solutions has emailed in an interesting piece of hardware he found at Astricon. Thanks Gerald!

"As I was walking though the booths at this year' Astricon conference, an interesting looking red box caught my eye.
The fonebridge T1/E1 Ethernet Bridge by Redfone Communications, LLC is a 4-port T1 to Ethernet gateway that connects T1/PRI circuits to Asterisk via TDMoE. This offloads the CPU burden associated with T1 connections and allows for sharing T1 circuits across multiple Asterisk servers, thereby increasing redundancy for mission critical applications. By offloading the T1 interface to a dedicated piece of external hardware, IRQ issues are eliminated, which would simplify installation when used with uncertified Asterisk Server hardware.

The only downside I can see with this device is that it does not come in a one or two port configuration. The MSRP is not too bad at $2,495 if you need a 4-port solution. But its 4-port configuration and price-point would limit its deployment to larger Asterisk installations.

Overall, the foneBridge looks like a winner for its intended audience.

Additional information about the fonebridge can be found at:

Announce: New Bug Marshal

Keving Flemming made the follow announcement about the newly appointed Bug Marshal:

"BJ Weschke has joined our bug marshal team and will be helping with code reviews and other types of bug support... so please welcome him and give him lots of work to do!

Can Skype Plug'n Play with Asterisk PBX ?

I found this today search around for more useful info about Asterisk:

Can Skype be a gateway to a PBX? Skype 2 PBX?

Here is a scenario posed to me by a Canadian company.

A company has four offices each with a local PBX. These PBXs are interconnected via SIP. (Of course the PBXs could have been interconnected with Skype, but that would be a boring story.)

This company would like to have any remote Skype Client have access to the corporate telephone infrastructure and as well, have any phone connected on any PBX access the Skype infrastructure, i.e. receive SkypeIn calls, place SkypeOut calls and place calls to any Skype Client. As well, all voice mail would handled by the PBX. When the remote client (e.g. a remote employee in San Diego) is unattended, all incoming calls are to be Call Forwarded so as to terminate at the PBX.

Here is a possible solution-

Click Here for Full Article

Wednesday, October 12, 2005

Release: Broadcast v0.9 | A Reliable Broadcast Application for Asterisk

Another release came across the User List a few mins ago. Hopefully we will see more stuff come out as Astricon moves on. Good work!

Broadcast is an Asterisk ( application which you may use to send a generic message over TCP/IP to any number of computers running software configured to listen for these types of messages. Being written in C, Broadcast will be dynamically loaded onto the Asterisk program on startup, making it a highly reliable and scalable option when compared with other solutions based on the Asterisk Gateway Interface (AGI) system...

Click Here for More Information about Broadcast

Click Here to Download v0.9.0

Hope someone finds it useful!


Monday, October 10, 2005

QueueMetrics - A call center monitoring software for the Asterisk PBX

With over 60 quantitative metrics available, your call center will run smoothly and problems will easily be pointed out. You can measure budget targets, SLA targets, agent activity and more with a level of details that goes down to listening to any call on any queue - right from your browser.

You can see taken calls, lost calls, agent sessions and call center activity, broken down by period, queue or agent.

QueueMetrics lets you see your call center in real time - what your agents are doing, how many calls are waiting for each queue, how much time has been spent in each stage of call handling.

QueueMetrics supports an unlimited number of inbound queues, each with multiple prioritized agents, and supports virtual queues made up of aggregations of existing queues.

QueueMetrics eases the burden of your agents too - they can easily see whether the're logged on or not, they can see their incoming calls as they are processed and they can even launch external webapps - passing data gathered from IVRs or Caller*ID information - at the click of a button.

Click Here for More Information

Click Here to Download the Latest Version (0.9.6)

Click Here to Download the User Manual

Editor Note: New Asterisk Book by Packt Publishing

Note: Today I have received an advance copy of this book from the very helpful Mohan Rapheal at Packt Publishing. Over the next week I will be reading this book and writing a review for AVN. If there is any questions you might want answered in my review please email me or post in the comment section I have opened for this and I will try and include your question with an answer. I am excited to see another book spreading the knowledge of Asterisk and I hope to see more to come in the future.


Announce: Fredericksburg ZPUG Meeting will have an Asterisk Flavor

Please join us October 12, 7:30-9:00 PM, for the fifth meeting of the Fredericksburg, VA Zope and Python User Group ("ZPUG"). Learn about Python configuration of Asterisk, an open source VOIP! Free food!

Rob Page, Zope Corporation CEO and President, will present a technical session on Asterisk [1] installation, configuration and operation. A brief discussion of connections to the public telephone network and internet telephony providers will be presented.

Hadar Pedhazur, Zope Corporation Chairman of the Board, will present a technical session on call handling and processing using Python extensions to Asterisk.

We will also serve delicious fruit, cheese and soft drinks.

We've had a nice group for all the meetings. Please come and bring friends!

We also are now members of the O'Reilly and Apress user group programs, which gives us nice book discounts (prices better than Amazon's, for instance) and the possibility of free review copies. Ask me about details at the meeting if you are interested.

General ZPUG Information
When: second Wednesday of every month, 7:30-9:00.

Where: Zope Corporation offices. 513 Prince Edward Street;
Fredericksburg, VA 22408

Parking: Zope Corporation parking lot; entrance on Prince
Edward Street.

Topics: As desired (and offered) by participants, within the
constraints of having to do with Python.

Contact: Gary Poster (

Wednesday, October 05, 2005

Astricon Update: 7 Days Left!!!

Astricon 2005

The Golden Asterisk Pub:

Open October 13 Great news for the Asterisk community: Digium will again host the all-conference party at AstriCon 2005. The event will be held at J.T. Schmidt's Brewery and features free micro-brewed beers, ales, and laggers. Busses will shuttle between the Hyatt and the brewery all evening, starting at 7:00 PM. The brewery will also provide a number of tasty snacks. So show up, drink a free beer, and talk telephony with the Asterisk community.

The Code Zone: Code In Style at AstriCon

Coders, don't forget your notebooks (or your desktops, or even your servers...) as AstriCon 2005 features "The Code Zone". The Zone is a room with everything you need to succeed with you Asterisk development project: internet access, test hardware power, pizza, Red Bull, and more. The Zone will also feature an Ask The Guru session with several of the leaders of the the community.

O'Reilly To Provide Asterisk: The Future Of Telephony To AstriCon Tutorial Attendees

O'Reilly, publisher of many technical books including the most extensive collection of open source guides on the market, has agreed to give away 500 copies of Asterisk: The Future of Telephony by Jim VanMeggelen, Jared Smith, and Leif Madsen (all speakers at AstriCon 2005). The books, which is a complete guide to installing and running Asterisk, will be given to the first 500 to purchase a ticket which includes the Asterisk Tutorials.

The Asterisk Open Source Showcase

The exhibit hall at AstriCon has a welcome addition this year - the Asterisk Open Source Showcase. The Showcase is a group of booths dedicated to open source projects which extend Asterisk. Stop by the showcase to see Asterisk running on a Linksys wireless router, on Soekris embedded systems. Check out the progress with AsterNic, the Flash Operator Pannel. See displays from some of the projects which are helping to extend Asterisk. The showcase will open with the exhibit hall on Wednesday, October 12 at 5:30 PM.

Click Here for Current News

Announce: New astGUIclient/VICIDIAL version released 1.1.7

Editor: Fresh off the press...Enjoy!


We've released another update to our Asterisk GUI Client suite: 1.1.7

Download Here

The client suite runs on Windows, UNIX and Mac, includes the astGUIclient client-side web app which extends your phone's functionality and the VICIDIAL client-side web app auto-dialer. This package is free as in 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 revision, we have finished our internationalization framework, introduced HotKeys key-binding into VICIDIAL and made the server-side apps more compliant with Asterisk 1.2.

As of this release, all client web-apps and administration pages are available in English and Spanish, with rough translations of French, German, Italian, Portuguese and Greek for the client web-apps only.

Check out the project blog for screenshots and more information:

Let me know what you think.



Tuesday, October 04, 2005

News: Sprint Nextel sueing over VoIP patents

Sprint Nextel is sueing vonage, voiceglo and for infringing on VoIP patents. Sprint Nextel claims to have about 100 patents on VoIP technologies. Does anyone know which ones this article is talking about, and if so does asterisk have any of those features?

The reason I am asking is that the article is vague, Vonage uses a fairly standard codec set, I dont know about the others. So if its not codecs I wonder if its something so generic that the patent would be tossed out upon challenge.

Anyone thinking about doing a VoIP business may want to get more info before proceeding since they may not have the millinos vonage has to fight this.

Click Here for Full Story

Bret McDanel

Announce: Voice over IP Directory Services

Hi All,

I am sure many of you are aware that by using VoIP devices one can make peer-to-peer calls. By devices I mean software phones, hardware phones and Asterisk. This feature is available, out of the box, in majority of devices today. Peer-to-peer calls are
'free' as in 'free beer' because they don't go through the service providers. All that is required to make a peer-to-peer call is 'peer connection information' of other user.

To make this global, where any VoIP user could make peer-to-peer call to any other VoIP user, we need the following a central repository which stores peer connection
information of all users an easy way to search and retrieve peer connection information of other users.

Voice over IP Directory Services (voipDS, pronounced as 'voip' D S) precisely addresses this need. Its a central repository that stores peer connection information of all users and also provides a search mechanism by which one could search for other VoIP users.

Searching and registering 'peer connection information' can be done manually via web based interface or automatically by implementing 'voipDS protocol'.

For more information, please visit,

voipDS is an 'Open source' effort and all the code will be released under GPL. If you wish to take part in this effort, please visit the website and join the mailing lists.

Feedback, comments and suggestions, please send to
"feedback at voipds dot org".

Thanks for your time,
Balaji NJL

Monday, October 03, 2005

Announce: Project Launched is pleased to announce a fork from the Asterisk software PBX. The software PBX builds on the solid foundation created by the developers of Asterisk.

The community plans to develop a robust offshoot from Asterisk building on its strengths, flexibility and user community. Some of the planned features include modular architecture, native support for Sangoma TDM cards, integrated faxing and eventually integrated messaging. will be community driven and released under the GPL.

Call for Developers


The project needs your help. We need testers, scripters, 'C' coders and bug marshalls. Lines are forming now, operators are standing by.

If you enjoy programming challenges and Voice Over Internet Protocol is the project for you. Project code will be licensed under the GPL.

Visit us today on the bug tracker, or IRC #openpbx, or download the code and have a look around. We value your skills, time and contributions.

Initial release of is slated for October 10, 2005. for additional information visit

Announce: Employment Opportunities at Digium

I was browsing the Digium site and came across there employment page. If you fit any of these jobs email digium. Enjoy

-Software Development Engineer
-Engineering Tech Technician
-Sustaining and Compliance Engineer
-Embedded Software Engineer
-Website Production and Development
-Software and Hardware Sales

Click Here for Full Job Descriptions


-College degree in related field preferred Knowledge and understanding of sales and marketing through distribution and VAR channels
-Knowledge of secondary languages, especially Spanish, a plus
-Familiarity and experience with telecom hardware sales a plus
-Understanding of networking, VoIP, and telecom a plus
-Familiarity with software sales, especially open source software and Linux OS a plus
-Familiarity with CRM and customer database tracking software a plus
-Must be in Huntsville, AL, or be willing to relocate

Send resumes to

Digium, Inc
HR Department
150 West Park Loop
Suite 100
Huntsville, AL 35806

Digium is an EOE employer.