News: Icasa licenses first Asterisk box

The first Independent Communications Authority of South Africa(Icasa) licence for an Asterisk-based PBX has been awarded to OpenVoice Holdings. The switching system (SWS) license means that OpenVoice's Asterisk PBX may be legally connected to the telecommunications grid.

The burgeoning Asterisk industry has been hard-hit by lack of Icasa licences, with reports of telecommunications monopoly Telkom refusing to connect lines to the Asterisk PBXs. This has resulted in some players misrepresenting their products to Telkom to acquire lines for their customers.

With the allocation of this SWS License, OpenVoice's Business Solutions subsidiary will be now focus on delivering its mainstream product -- VoIP PBX solutions -- to the public and will continue to follow its dual strategy of utilising a channel and direct delivery model.

Asterisk is an open source PBX software system that offers many of the benefits of a traditional PBX on standard PC hardware, thus significantly lowering the costs involved.

News: Asterisk Appliance Development

Ranch Networks has announced the development of security code for Asterisk, the open source VoIP project, that allows enterprises to combine Asterisk with Ranch appliances. The code is certified by Huntsville, Ala.-based Digium and is available in the latest Asterisk version, number 1.2.3.

The code uses a standard developed by the IETF called Middlebox Communications to open firewall ports only when a call is actually in progress. This should make enterprise firewalls far more secure.

Ram Ayyakad, CEO of Ranch, calls this "dynamic firewall control", and it is already a part of the latest Asterisk core, v. 1.2.3. It will also be available as an integral part of the next major Digium release. For the moment, Ayyakad admits, "we are still working with Digium to clarify a few things. The challenge was to put our code into the core of Asterisk."

The benefits are numerous, Ayyakad says. "It's really good bulletproof security, hardware based queuing, the ability to segregate data and voice, bridging, and there's no proprietary thing going on."

Click Here for the Full Article

Help Article: Echo Cancellation on Asterisk

Note: My name is Matt Birkland, I work as a VoIP Engineer for VoiceIP Solutions an Asterisk Provider in the Seattle area. Every Monday I will be submitting a one page Asterisk/VoIP tip of the week on the blog. This week we will discuss echo cancellation.

Echo can stem from several sources. Echo can commonly be generated by the digital to analog conversion. This is mostly problematic on POTS (Plain Old Telephone Service) lines. Mainly through (in my experience) acoustic echo is the culprit. Such as when your voice on the receiving ends speaker is also picked up by the receivers mic. This can be tedious to configure on a phone by phone basis, so I usually turn on aggressive echo cancellation in Asterisk.

Aggressive echo cancellation can only be enabled by editing zconfig.h located in the zaptel source directory. I'm assuming the zaptel source library is in /usr/src . By the way, cleanly shut down Asterisk before following the rest of this guide. Observe:

1) Change directory to zaptel source directory.
example:
[matt@localhost ~]$ cd /usr/src/zaptel

2) I'm using nano to edit the file, but pico, vi, emacs, or any text editor will do.
example:
[matt@localhost zaptel]$ nano zconfig.h

3) Below highlighted in red is the section of the file that need editing to enable aggressive echo cancellation.
example:
* Pick your echo canceller: MARK2, MARK3, STEVE, or STEVE2 :)
*/
/* #define ECHO_CAN_STEVE */
/* #define ECHO_CAN_STEVE2 */
/* #define ECHO_CAN_MARK */
#define ECHO_CAN_MARK2
/* #define ECHO_CAN_MARK3 */

/*
* Uncomment for aggressive residual echo supression under
* MARK2 echo canceller
*/
/* #define AGGRESSIVE_SUPPRESSOR */

4) The next step is to erase the delimiters. For this configuration file the delimiter is "/*" or forward-slash and a Asterisk.
before example: /* #define AGGRESSIVE_SUPPRESSOR */
after example: #define AGGRESSIVE_SUPPRESSOR

Do not erase the '#' sign!!!!! In BASH and many config files it is common to use '#' as the delimiter, but not in this case!

5) The last step is to recompile the zaptel source code. Then restart Asterisk and enjoy.

-Matt
http://www.voiceIPsolutions.com

Announce: New GXP-2000 Beta firmware available

From the usual place:
GXP2000 Beta Download

Note, there are two (and it took me a bit of a while to figure out) images to be loaded. Copy the ...a.bin's and the .bin's to your http provisioning directory, and reboot. The phone _must_ load the .bin files before it understands the ..a.bin files.

After it loads the first one, the phone does lock up with the 'Grandstream' logo displayed. I left it sitting there for a minute just to make sure it wasn't flashing itself or anything, then power cycled it. Then it requested the a.bin files, and away it went.

For those that didn't see the unofficial beta firmware, these phones now support BLF and call pickup (but, in a very asterisk-centric way). Use your standard HINT to get the BLF, but when the user pushes the flashing button, the phone sends '**[xtn]'. So you want something in your dialplan like:

Exten => _**XXX,1,Pickup(${EXTEN:2})

--Rob

Release: Asterisk 1.2.4 and Zaptel 1.2.3

Asterisk 1.2.4 and Zaptel 1.2.3 have been released!

This update of Asterisk includes a fix for a significant memory leak in the expression parser that is present in all previous releases of Asterisk 1.2. This version of Zaptel includes support for the new generation of VPM100M echo cancellation modules from Digium. For further information about all changes that have been made, consult the appropriate ChangeLog in the tarball or on the ftp site.

Thank you!

-- The Asterisk Development Team


Click Here for More Information

News: Small Businesses Place 10 Million Calls With Fonality PBXtra Phone System

Fonality, the leader in affordable IP-PBX systems for small businesses, today announced that its customers have placed 10 million calls using Fonality's PBXtra phone system. PBXtra, the world's largest distributed deployment of Asterisk(TM), became available in October 2004 and is used by hundreds of businesses with thousands of lines in operation. More than 800 resellers in 76 countries are offering PBXtra to small businesses looking for an enterprise-class phone system at an affordable price.

"Fonality is the leader in the Asterisk-based PBX market," said Chris Lyman, Fonality's founder and CEO. "The reason is simple -- PBXtra provides small businesses the phone system they always wished they could afford and the ability to use it without going to phone school. If you can use Hotmail, you can use PBXtra."

Click Here for the Full Release

Release: Astribank-8 available to public

The Astribank-8 was initial annouced at this year's Astricon. Xorcom has finally added details on how to purchase.

Features:
* 8 FXS ports
* 2 relay output ports for peripheral devices (alarm, gate, etc.)
* 4 input ports for peripheral devices
* USB-2 direct connection to Asterisk server - no PRI card required
* Full support of Asterisk conference capability
* High-quality analog telephony
* "Plug-&-Play" installation using Xorcom Rapid
* Caller ID, message waiting indicator, individual ring patterns
* Supports international impedance and all analog telephone models
* Indicator lights
* USB "Plug and Play" operation
* Automatic configuration and setup with Xorcom Rapid distribution

Announce: AstAutoDialer 0.5 released

What is AstAutoDialer?

* A desktop application for bulk auto dialing from Asterisk
* Supports dialing on multiple lines simultaneously
* Call scheduling
* Runs on Windows
* You don't need Asterisk server to install it
* But you do need a Asterisk server if you want to dial out
* Includes a graphical dial plan designer, AstPlanDesigner
* Not open source but free for use

Click Here to Download

Network World Magazine Names Mark Spencer of Digium Among 50 Most Powerful People in Networking

The original creator of Asterisk and pioneer of open source telephony, announced today that President Mark Spencer has been named one of Network World Magazine's 50 Most Powerful People in Networking. The annual list featured in The Power Issue of Network World's Signature Series takes a look at the people, companies, technologies and ideas that are transforming the networking industry. The Power People list profiles 50 of the most aggressive, competitive, masterful and thought-provoking individuals of 2005.

"At 27, Spencer has the youth and momentum to set the network industry spinning. He became famous in 2005, as the industry took notice of Asterisk, the open source PBX system he created, and of his continued pioneering work in open source telephony. Not only is Asterisk an intriguing open source option for VoIP, but as an open source hardware product it has become the proving ground for the entire open source movement. Spencer has been as admired for his marketing skill as for his technical abilities," stated the article in Network World.

The publication chose the recipients based on nominations submitted by its editorial staff, columnists, newsletter writers, Lab Alliance members and external sources. Unsolicited nominations were not accepted for consideration and decisions were based on criteria that include revenue, market share, customer service, leadership and industry influence.

"Being mentioned along side of Gates, Chambers, Ellison, and McNealy is pretty cool, but it really is the entire Asterisk community that has made the platform what it is today," said Mark Spencer. "Asterisk has long left the garage and is being fostered by talented developers around the world, many of which work at the networking and telecom powerhouses. Its use is more widespread within those communities than people realize, so piquing the interest of large networking vendors is really just a natural progression."

Click Here for the Full Article

Announce: Asterisk at Home 2.4 released

The latest version of Asterisk@Home has been released which includes bugfixes. Click here to Download the latest version.

Announce: iDefisk 1.30 released (Iax Client for Mac)

We tried to fix some of the issues found in the field test and just released a new version available for download.


This fixes the problem with the audio devices, (a bug we were hunting for some weeks now), puts the library in the correct directory, uses a nice installed and has a more mac style configuration menu.

Time to start working on zeroconf and universal binaries!

Greetings,

Zoa

Announce: New Grandstream GXV-3000 H.264 Launched at Internet Telephony Conference

The new Grandstream GXV-3000 H.264 based SIP video phone launched at Internet Telephony Conference & Expo yesterday. I also spoke about this during my chit-chat with Digium's Mark Spencer. Based on completed interoperability tests conducted by engineers from Grandstream and its partners, GXV-3000 works seamlessly with SIP soft client from Counterpath, IP-PBX or softswitch products from Digium, Netcentrex, and Pingtel.

Click Here for the Full Article

News: Skype-to-Asterisk(SIP): Progress - Part 1

Click Here for Skype-to-Asterisk(SIP): Progress - Part 1

Annouce: Sangoma Releases the New A200 FXO/FXS Card

Meeting growing global demand in the telephony space, the A200 and Sangoma's REMORA system together, comprise the FXO/FXS version of the company's Advanced Flexible Telecommunications (AFT) hardware designed for optimum support of analog voice traffic.



"There is still a huge demand and need for FXO/FXS technologies in many countries around the world," says Sangoma Technologies President and CEO David Mandelstam. "Not all telephone systems require the latest VoIP technologies, but still require robust analog capabilities to maintain quality of service. We understand this need and are taking an industry lead by providing a price/performance solution that our competitors will now follow."

The A200 solution supports any combination of up to 24 FXO or FXS connections. A single PCI slot host connection for all ports ensures common synchronous clocking for all channels. The base AFT architecture is shared with Sangoma's A101, A201 and A104 and soon to be released A108 cards ensuring common 3.3V/5V, high performance PCI compatibility.

In keeping with the philosophy of providing complete flexibility to all customers, the A200 has field upgradeable firmware to take advantage of hardware and software improvements as they become available. In continuing to raise the bar by providing price performance for softPBX solutions, Sangoma is shipping an optional low density HW based echo canceller for the A200 series.

The A200 consists of a REMORA daughterboard mounted on the AFT PCI card. The REMORA card has two sockets each of which can accept a FXO-2 or FXS-2 module. Each FXO-2 or FXS-2 module supports two FXO or FXS ports respectively.

Up to five additional REMORA daughterboards can be mounted in empty slot positions beside the A200 assembly connected to the A200 by a backplane bus connector.

Click Here for More Information

Announce: Snom 360 with integrated XML Objects

Dear User,

the new snom 360 is able to use services from standard web servers. Users can deploy customized client services with snom 360 and interact with other users via the keypad. The snom 360 will use HTTP protocol from standard web servers, like Apache.

Typical Services:

~ 1. To-do lists
~ 2. Stock Information
~ 3. Weather
~ 4. Provisioning
~ 5. Agenda
~ 6. Telephone directory


For further information go to: Snom Information

Note: *That is a pre-release, probably the software is still unstable*

Best regards,

Hirosh Dabui

News: Open Source Pioneer Larry Augustin Joins Fonality's Board of Directors

Fonality, the leader in affordable IP-PBX systems for small businesses, today announced that Open Source pioneer Larry Augustin has joined its board of directors and will assist the company with its corporate strategy in the Open Source IP telephony market. Fonality's PBXtra IP-PBX is based on the Open Source Asterisk platform and, since it began shipping in October 2004, has placed millions of calls and has been deployed by hundreds of businesses with thousands of lines.

Click Here for the Full Release

Update: No audio - Update your Asterisk

This morning we discovered a serious bug that stopped all bridged audio in our Asterisk servers. Mark found the problem and soon fixed it.

If you get this problem today, please update your Asterisk server. A fix has been commited to the subversion repository for 1.2 as well as trunk.

A fixed 1.2.3 release will be published on ftp.digium.com as soon as we can find a release engineer (consider the time zone problem).

A big thank you to everyone in the IRC channel that helped us locate this issue and to Mark that fixed it so quickly.

/Olle

FTP Download

Announce: Call Accounting Mate Available

Call Accounting Mate is an fast and reliable call accounting software package. Call Accounting Mate could be deployed in virtually any enterprise including retail, government agencies, brokers, professional firms, banks, hospitals and universities. Telecom mangers find it increasingly difficult to allocate telecom costs to various departments or cost centers or individuals. This solution will pinpoint charges, highlight misuse and increase productivity.

Click Here for a List of Features

CallAccounting

News: Signate Announces Hosted Telephony System SigPRO at ITEXPO

By: Tom Keating

Signate is announcing SigPRO. SigPRO is a hosted telephony system for telephone service providers with 5,000 to 500,000 customer extensions that Signate claims "sets a new price/performance standard".

I have heard various issues surrounding scalability issues when it comes to the Linux-based Asterisk IP-PBX solution, so I contacted one of my sources and asked about Signate's scalability claims. My source responded, "Signate is in for a shock when they really try to scale this stuff. They are using GPL'd code from Asterisk, and have not licensed the code. This means that they are likely not changing core Asterisk source code to implement scale. This means they are in for a bit of a surprise when they actually start getting hundreds, let alone hundreds of thousands concurrent users on one box."

Click Here for the Full Article

News: Asterisk's Mark Spencer Speaks

The keynotes at Internet Telephony Conference & Expo keynotes kicked off today, the first day of this four-day event, with Mark Spencer of Digium/Asterisk to a huge audience. Mark started his talk by saying that his company runs VoIP on a network that is the worst case scenario and it allows them to find out if products are broken before others find problems. As developers they are thrilled to do this but not everyone in the company is as excited by this strategy (for obvious reasons).

Spencer says he has to be sensitive to customer calls as people associate telephone quality with the products you make. Mark mentioned that his phone system was recently inundated by callers looking for a very funny message (Mark indicated the sound clip is available on CD and did not explain how you could hear it if you missed the conference).

Click Here for Full Article

Nerd Vittles: As Easy As 1-2-3: A Telephone Reminder System for Asterisk

Nerd Vittles today introduces a new Telephone Reminder System for Asterisk:


"The reminder system is actually quite simple to use. You dial extension 1-2-3 on your Asterisk system, enter your password, and then you'll be prompted to record a reminder message. Next you enter the phone number, date, and time for delivery of the reminder message. When the appointed date and time arrives, Asterisk will place the call to the number you specified using your default dialing rules and will play the customized
reminder when the call is answered. If the call is not answered, the call will be repeated n number of times with a delay between calls of x minutes before giving up on the call. You'll get an email with the call reminder setup if desired. You also get to configure the number of retries and the delay between calls."

Click Here for the Full Article

Digium, Ranch Networks Optimize Asterisk Firewall, Bandwidth Allocation

Open source pioneer Digium has incorporated firewall security and bandwidth allocation to its latest release, version Asterisk 1.2.2, through a year-long collaborative effort with data network security specialist, Ranch Networks, which is also unveiling a series of hardware appliances optimized for the open source PBX.

The security/performance enhancement is designed to help resolve a central issue with voice-over-IP: trafficking the firewall. "We came up with a product that can dynamically open and close holes in the firewall that is entirely asterisk-controlled," said Ram Ayyakad, co-founder and CEO of Ranch Networks.

Click Here for Full Article

Release: Ekiga 2.00 BETA 1 is finally there!

Note: This is the beta of a new softphone client that supports SIP and H.323.


After more than one year of development, Ekiga 2.00 BETA is finally available.

Among the features, you can find:

* Call Forwarding on busy, no answer, always (SIP and H.323)
* Call Transfer (SIP and H.323)
* Call Hold (SIP and H.323)
* DTMFs support (SIP and H.323)
* Basic Instant Messaging (SIP)
* Text Chat (SIP and H.323)
* Possibility to register to several registrars (SIP) and gatekeepers (H.323)
* Possibility to use an outbound proxy (SIP) or a gateway (H.323)
* Message Waiting Indications (SIP)
* Audio AND Video (SIP and H.323)
* STUN support (SIP and H.323)
* DTMFs support
* LDAP support

Click Here for More Information

Announce: PodMail 1.0 (GPL)

Hello Asterisk Community.

While sitting at lunch the other day I had a typical napkin-prototype idea:
What if I could make my Asterisk Voicemail accessible as a Podcast in iTunes? Three hours later with the help of two friends I had a working proof of concept. Now we are releasing the polished version of this idea as PodMail 1.0.

PodMail brings together open-source telephony and Podcasting to create a new, useful way of accessing voicemail and podcasting.

PodMail integrates with Asterisk to provide a secure podcast of your voicemail. Supporting authentication directly against voicemail.conf or using an LDAP directory, PodMail allows you to subscribe to your own voicemail box. Each time you dock your iPod, your new voicemails will sync right along. Listen to your voicemail at your convenience and without using cell minutes.

PodMail also allows for a brand new type of PodCasting. Unchain Podcasting from the computer! Configure PodMail for public access and you have a ready-to-run PodCast. Updating your Podcast is as easy as phone call. Moblogging has never been so easy or flexible.

Live Demo:
Do not miss out our live demo at http://podmail.alkaloid.net/
Leave us a message in one of our mailboxes, subscribe to one of the PodMail Podcasts, then see and hear your message immediately!

Check out the PodMail Documentation and Installation Notes at:
http://projects.alkaloid.net

PodMail is released under the terms of the GPL.

Enjoy!
/BAK/

News: Mark Spencer Interview

Greetings!

On January 19, 2006, I featured VoIP and open source telephony pioneer, Mark Spencer, on my podcast, Technology & Coffee. To listen to this interview visit:

http://www.ronaldlewis.com/coffee

Also, Tom Keating, CTO and VP at TMC Labs, has blogged about it as well.

http://blog.tmcnet.com/blog/tom-keating/asterisk/
mark-spencer-podcast-interview.asp
http://blog.tmcnet.com/blog/tom-keating/voip/
gabcast-audio-blogger-service.asp

Regards,

Ronald Lewis
Denver, Colorado
www.riverscapecorp.com
www.ronaldlewis.com

Release: New astGUIclient/VICIDIAL release: 1.1.9

Hello,

We've released another update to our Asterisk GUI Client suite: 1.1.9

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 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 focused on several security enhancements as well as fixing bugs and several new features like a favorites panel for astguiclient that will show realtime extension state and a Scripts tab in vicidial that will show a script to read with customer data filled in. We have also tested the suite on Asterisk versions through 1.2.2

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

Check out the project blog for screenshots and more information:
http://astguiclient.blogspot.com

Let me know what you think.

Thanks,

MATT

News: Asterisk Development and Release Cycle

Asterisk 1.2 was released over 1 year after Asterisk 1.0, which resulted in many users trying to run the development version of Asterisk in a production capacity so that they could take advantage of the new
features that had been added. This produced a flurry of extraneous bug reports and caused extra work for the developers as they could not work on changes that would actually cause disruption of the development tree.

In an effort to combat this problem, and to give the community a more predictable release cycle, the process is being organized so that such a long time between releases will never happen again.

Beginning in January of 2006, we will produce new major Asterisk releases on a six month cycle.

The development cycle will be organized in this fashion:

MONTHS 1 - 3

The first three months of the development cycle are when the development branch will be changed most drastically. The tree is open to large architectural changes as well as new feature enhancements and bug fixes.

MONTHS 4 - 5

For the next two months, the development branch will no longer receive architectural changes. New features that are ready to be merged will still be accepted at this point.

MONTH 6

The last month is reserved for beta testing. No more features will be accepted for the upcoming release. Beta releases will be made on a weekly cycle, culminating in one (or two) release candidate releases just before the final release.

Asterisk 1.4 is scheduled to be released in the beginning of July, 2006. Once the release is made, a branch will be created. This branch will then receive maintenance for bug fixes only. At that point, the development cycle will start over to prepare for the next major release of Asterisk, scheduled for January of 2007.

The Asterisk Development Team

Announce: Asterisk::LCR released on CPAN

After a few extra days of hard work, debugging, and many coffees, I am pround to announce that Asterisk::LCR has been released on CPAN.

Asterisk::LCR is an open-source, Perl-based collection of tools to help you manage efficiently multiple VoIP providers with your Asterisk installation.

It is capable of importing providers rates from multiple providers, comparing these rates, and generating optimized Asterisk dialplans.

It's a /command line tool/. Hence it is designed for system administration and people with a minimum of technical know-how.


Features:

- Capable of comparing rates in different currencies and with different billing schemes (connection charge, 30/6, 1/1, etc)

- Pluggable comparison system (in the future, I intend to write a module that also takes into account the ASR of a given route to weight its cost)

- Pluggable importer system (so we can add more VoIP providers in the future). Currently supports VoIPJet, Nufone, and PlainVoIP. Others providers are invited to contact me if they want to be added to this list.

- Pluggable dialing strategies. At the moment there is MinCost (dial from the cheapest to the most expensive providers sequentially) and MinTime (dial the $n cheapest providers simultaneously to minimize post dialing delay).

- Capable of translating prefixes so that you can generate dialplans using different dialing locales (for example, using french dialing conventions along with US providers)


I'm sure that they are many quirks and bugs to sort out. You can check the documentation here.


perl -MCPAN -e 'install Asterisk::LCR'


Let me know how it goes!

Cheers,
Jean-Michel.

News: Aheeva Releases New Version of Asterisk-Based IP Contact Center Suite

Six years ago, Aheeva was a just a good idea. Today, announcing Version 2.0 of its Asterisk-based IP contact center software, the small but already global provider of customer relations and contact center solutions attracts notable partners including Digium and now SugarCRM.

Aheeva Contact Center Suite Version 2.0 and SugarCRM
Aheeva CCS Version 2.0 is smarter IP contact center software, featuring improved quality monitoring tools, enhanced statistical analysis and tools to increase agent efficiency. Now supporting SugarCRM, a customer relationship management solution, Version 2.0 remains a cost-effective solution based on open source technology.

Aheeva's business communication solutions are fully-customizable and integrated on a single platform. Aheeva CCS Version 2.0 requires only a Web browser for access, making remote monitoring, management and analysis possible anywhere with an Internet connection.

Click Here for Full Release

Announce: Asterikast release 1st HowTo video on setting up the Asterisk PBX

Asterikast is a podcast that teaches and explains about the Asterisk PBX by Digium. We also plan on having videos that can help step you through the process of setting up your very own Asterisk PBX. Check back for new updates. We will also have rss feeds for your use as soon as our first episode is recorded.

Click Here for More Information

Announce: AsteriskWin32 0.56 released

AsteriskWin32 Features:

Asterisk-based telephony solutions offer a rich and flexible feature set. Asterisk offers both classical PBX functionality and advanced features, and interoperates with traditional standards-based telephony systems and Voice over IP systems. Asterisk offers the features one would expect of a large proprietary PBX system such as Voicemail, Conference Bridging, Call Queuing, and Call Detail Records.

Click Here for More Information

Click Here to Download

Release: Asterisk 1.2.2 Released

Greetings everyone!

The 1.2.2 versions of Asterisk, Zaptel, and Libpri have now been released. The source tarballs are available for download on ftp.digium.com.

For details about what has changed:
ChangeLog for Asterisk
ChangeLog forZaptel
ChangeLog for Libpri

We are also excited to announce the release of a special version of Asterisk 1.2.2, called Asterisk-NetSec. It includes some very exciting features not available in any other version of Asterisk, or even any other related product! Please view the appropriate README and ChangeLog for more details.

Asterisk-addons and Asterisk-sounds will remain at version 1.2.1. Previously, all packages were updated to reflect a matching version number, even if no changes have been made. From now on, releases will only be made when changes have actually been made. Even if version numbers do not match, it is safe to use all of these releases together, as long as all of them are the latest version available.

Thank you!

News: 50 open source apps rolled into one server

Note: Here comes one of the first of many all-in-one boxes that will include Asterisk. Enjoy!


Portugal may not be well known as an IT hotbed, but it's the source of edgeBOX, the latest entrant into the increasingly competitive business gateway market. With a starting price of £1500 for a 20 user system, edgeBOX rolls some 50 open source apps plus proprietary management tools into an all-in-one device aimed at smaller businesses and the service providers who sell to them.

The company behind it, Critical Software, claims that the device does everything a business might need, from a wireless router, through web, e-mail and DNS/DHCP services, to IP telephony - it includes the Asterisk IP PBX. It's also a firewall with VPN, authentication and anti-spam/anti-virus capabilities, all on a Linux base, of course.


Click Here for Full Story

[Nerd Vittles] Deploying Voice Over Wi-Fi with Asterisk

Nerd Vittles today presents a comprehensive HOW-TO on building and deploying a Voice Over Wi-Fi Network with Asterisk:

Getting Wi-Fi to play nice is another one of those grammatically incorrect, but thorny Linux problems that Asterisk inherits because of its roots. In the Windows and Mac worlds, we've become accustomed to plug-and-play for things like Wi-Fi USB sticks, but it ain't that easy with Linux unfortunately. Once you get the right device, this project will take you less than 30 minutes to complete. But we've invested almost a week getting everything to work ... so you won't have to. If the Linux community ever wants to see Linux used as a desktop PC, this piece of the puzzle needs some work!

Deploying Voice Over Wi-Fi with Asterisk

Switchvox Receives INTERNET TELEPHONY(R) Magazine's "Product of the Year" Award for 2005

Switchvox, a leading provider of IP PBX phone systems for SMBs, announced today that Technology Marketing Corporation (TMC(R))'s INTERNET TELEPHONY(R) magazine (www.itmag.com) has named Switchvox as a recipient of a 2005 Product of the Year Award. INTERNET TELEPHONY has been the VoIP Authority Since 1998(TM).

Switchvox IP PBX systems offer SMBs a turnkey solution, delivering an intuitive interface and powerful features for a fraction of the cost of comparable legacy systems. By supporting both traditional telephony and Voice over IP, Switchvox allows businesses to choose what works best and saves them money. Switchvox employs open standard communication protocols to interconnect with products from many other vendors, which means customers will never be locked into a single vendor's solution, a problem commonly associated with traditional phone systems.


Click Here for Full Article

News: Florida Linux User Community(Flux) Meeting (Mark Spencer attending)

Hi all,

We're kicking off 2006 with a bang. Our first speaker for the year will be Mark Spencer, inventor of the Asterisk Open Source telephone PBX software!

Thanks to longtime FLUX member Mick Weiss, Mr. Spencer has agreed to speak to our group while he is in town late this month. For that reason the January meeting will be rescheduled to TUESDAY JANUARY 24th in the Nova Southeastern University medical school (directions).

We'll have a big auditorium because we expect a whole lot of people to attend!

Refreshments begin at 6:30pm, presentation starts at 7:00pm.

Click Here for More Information

Announce: AsterFax Beta 6 released

AsterFax is an email to Fax gateway for Asterisk.

You can download AsterFax at: http://asterfax.sourceforge.net

The newly released Beta 6 introduces a host of new features including:

Cover Page
The body of an email is now used as a cover page to any file attachments.

Self Diagnosis
AsterFax now attempts to identify any configuration problems and provide a plain english explaination on how to fix them.

Archive
Faxes can now be automatically archived to a mailbox

Delivery Receipts
A delivery receipt is now sent on the successful transmission of a fax.

Expanded set of supported file attachments

And many more.

Oh and of course lots of bug fixes too :)

Announce: New RPM packages for CentOS4.0

Greetings list,

It's been a while since I've been able to focus on asterisk packaging but this weekend I took some time to audit and recompile packages for CentOS 4.2. You can find them here.

ftp://ftp.linuxsys.com/ftp/pub/releases/CentOS-4.0

You have your choice of 1.2.1 or 1.0.10 releases. If you need zaptel modules then install this kernel as well:

ftp://ftp.linuxsys.com/ftp/pub/releases/CentOS-4.0/kernel

SRPMS are available for those wishing to recompile zaptel against their own kernel.

Features of this release
========================
- 1.2.1 patched with spandsp-0.0.2pre22
- 1.0.10 patched with spandsp-0.0.2pre21
- init script launches safe_asterisk by default
- compiled to include cdr_addon_mysql.so and format_mp3.so
- asterisk console is automatically launched on pseudo tty8
- zaptel init script configs are moved to /etc/sysconfig/zaptel
- tested to work with AMP (required software available as rpms)

Other packages released
========================
astcc-40-1.RHEL4.LSE.i386.rpm
asterisk-sounds-31-1.RHEL4.LSE.i386.rpm
gtkiaxyprov-17-1.RHEL4.LSE.i386.rpm
gastman-54-1.RHEL4.LSE.i386.rpm
iaxyprov-15-1.RHEL4.LSE.i386.rpm
lame-3.96.1-RHEL4.LSE.1.i386.rpm
lame-devel-3.96.1-RHEL4.LSE.1.i386.rpm
perl-DBD-mysql-3.0002-1.RHEL4.LSE.i386.rpm
perl-IPC-Signal-1.00-1.RHEL4.LSE.i386.rpm
perl-mime-construct-1.9-1.RHEL4.LSE.i386.rpm
perl-Net-Telnet-3.03-1.RHEL4.LSE.i386.rpm
perl-Proc-WaitStat-1.00-1.RHEL4.LSE.i386.rpm

As you should expect theses packages come with no warranty whatsoever but I would like some feedback so please feel free to contact me via email - amcrorylinuxsyscom.

Best Regards,

Andrew McRory - President / CTO
Linux Systems Engineers, Inc.
Located in beautiful Tallahassee, Florida

VoIP is the New POTS

Note: Interesting article, it blows my mind that 1 billion people are using the internet today.

Voice over Internet Protocol is about to become Plain Old Telephone Service. By every financial, operational and cultural metric, VOIP is on the way to replacing the switched circuit telephone system as the standard communications medium.

Consumers and businesses now expect services from their digital mobile telephones and the Internet that the old telephone system cannot provide. Investment is pouring into advanced IP technologies, not old telephone systems. As a result, the contest between switched circuit telephony and VOIP is almost over; VOIP will be the clear winner. The new battles will be between owners of infrastructure and VOIP service providers and among application developers.

Click Here for Full Article

How To: Building an Embedded Asterisk PBX

Note: A nice article on Tom's Hardware.


"Astlinux is a bundled distribution of the Asterisk open source iPBX private branch exchange (PBX) software and a Linux operating system. Originally developed by Mark Spencer at Digium, Asterisk is the leading open source software in the telephony/VoIP space. Asterisk excels at combining traditional TDM telephony capability - provided through hardware from Digium and others - with VOIP services. These include call routing, media gateway, media server and SIP signaling capabilities."

Click Here for Full Article

MINNESOTA: TwinCities Asterisk Users Group -Saturday 01/14/2006

Hello,

The next Asterisk Users Group meeting has been scheduled for tomorrow, January 14th at 11:30am.

Its a new year and it's been two months since our last meeting.

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

Sound Choice Communcations is located in Bloomington Minnesota, just 1/2 mile west of the Mall of America. The 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...

Google Maps Link

This month we'll take another look at the dialplan using extensions.conf and a look as what's now possible using the new AEL. Please come and share your own tricks and learn from others. As always, free food. If interest is small in reviewing dialplan tricks, we'll discuss QOS issues and solutions. (from our wiki wish list)

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.

VoIP-Info Link

Asterisk VoIP News 1 Year Anniversary!!!!

Editor Note: First Off I want to thank everyone that supports this resource be either sending my stories and information about Asterisk or visits this site regularly. You have helped build this from nothing into a vibrant site. Over the last year I have watch Asterisk really come of age and start to become a technology everyone has to talk about. Week after week Asterisk has penetrated into bigger media sources spreading the word.

Predictions: I think I can agree with most that 2006-2007 we are going to see a huge upswing on Asterisk installations and more unique applications where Asterisk can and will be applied.

Future For AVN: During the next year I am going to continue bring you interesting news about Asterisk and VoIP. Also I am going to increase the amount of unique content and stories about Asterisk. As always you don't see something on my blog that should be on here EMAIL ME!. I can't stress this enough, this site is our site.

Now on to the Fun Stuff

Statistics for AVN from the last 365 days:
Page Loads: 60,553
First Time Visitors: 42,128
Returing Visitors: 12,595
Most Popular Posts: Astricon Reports and Asterisk Case Studies
Number of Countries that has visited: 107 (Even Malta)
Top 5 Countries: USA, Canada, UK, France and Australia
Most Popular Resolution: Still 42.86% 1024x768(with 1280x1024 close behind)
Most Used O.S.: Well you know that one but Linux has a respectable 10.60% :)

Final Note: If you have a comment or insightful commentary I will be editing this post with the interesting emails I get.

Announce: AEL2 -- The Future --

Call to Action!

For those who have the courage/ability, go grab an SVN copy of the asterisk release, the HEAD version,

and my latest patch, from: http://bugs.digium.com/view.php?id=6021

Right now, the latest version of the patch is 0.10.

apply it to the SVN head version, and do a "make".

Read the Wiki on AEL2: http://www.voip-info.org/wiki/view/Asterisk+AEL2

Look at the examples at:
http://www.voip-info.org/wiki/view/AEL+Example+Snippets

Then, sit down and rewrite your extension.conf to /etc/asterisk/extensions.ael2

Use "utils/aelparse -n" to check your file. Get rid of all the syntax errors. Repeat until clean.

Then, see if your extensions.ael2 loads. Remove all the contexts except [general] from your extensions.conf, and restart asterisk. Test the new dialplan.

Now, at this point, you have some information that would be useful to me! I need to know your trials, troubles, confusions, and solutions. Perhaps there's some added check that AEL2 could make that might have warned or helped you. Perhaps you'll help find some bug and solify AEL2.

WHY ON EARTH WOULD I WANT TO DO ALL THAT WORK?, you might ask!

In answer to that, my reply is the Parable of the Programmer:
******
In the beginning, there was machine code, and the Programmer thought it was great.

Then, along came the assembler, and the programmer found it very useful.

Then, along came the macro assembler, and the programmer was excited indeed.

But then came the "programming language", and the programmer left behind the "macro assembler", and never went back to it.
******
Now, the goal is for the same to happen to you, in moving from the extensions.conf method of programming dialplans to extensions.ael2.

If you don't think it will improve the quality of the code you write for dialplans, or reduce your costs of dialplan development, then we'll scrap the project and hand ourselves over to public humiliation. ;^)

murf

Announce: Web-MeetMe v2.0.0

[New Features]
1. Added focus and tab-order to all input fields
2. Dynamic generation of date/month/year listboxes
a. It is no longer possible to schedule an invalid date.
3. Added 'Extend' and 'End Now' buttons to the monitor
page.
4. Invite button on the monitor page. This greatly simplifies the process of adding callers to a conference.
The ./lib/defines file includes definitions for the prefered channel and context
*****************
5. Call history report. Support for this feature requires the php script ./lib/cbEnd.php be running at all times. This also requires a new table in the meetme database if you're upgrading from an earlier release.
****************

[Location]
http://www.fitawi.com/Asterisk
[Files]
Web-MeetMe_v2.0.0.tgz (required)
app_cbmysql.c (required)
cbmysql.conf (required)
cb-extensions.conf (suggested)
README (suggested)

[Installation]
See the README

[Features]
1. Schedule new conferences
a. Control start and end times
b. Set conference pin #
i. Generate one if the requester leaves it blank
ii. Identify pin # conflicts (another conference with the same pin is scheduled at the same time)
c. Set Admin and User passwords
i. Generate a user password if an Admin pw is set but the User pw is blank
d. Weekly recurring conferences with the same settings
e. Select MeetMe flags per conference for Admins and Users
2. Email the details for a successfully scheduled conference
3. Separate views for Current, Past and Future conferences
4. Ability to modify the end time of a running conference
a. Can also reschedule a past or future conference.
5. Monitor realtime conference activity
a. Mute/Kick participants
6. Optional authentication
a. Currently Active Directory or LDAP based
b. Authentication is abstracted so unix/PAM/DB/RADIUS
support could be easily added
7. Users can only monitor, update or delete their conferences
8. Verified administrators can monitor, update or delete any conferences.
9. Updated to Asterisk 1.2.0
a. Changes to the Manager interface may have caused
support for 1.0.X to slip, I cannot test that)

Thanks and enjoy,
Dan

***Beta testers and anyone who downloaded v2.0.0 before today****
The only changes from the beta was a cosmetic change to work with non-IE browsers and a couple of installation hints. I only received feedback from one tester, so it appears the package is ready to go.

***Developer help/guidence request***
The PHP script to monitor conference endtime and up date the CDR is fragile. If Asterisk is shut down for more than 30 seconds, the script exits. I'd like to make it more resilent. If any PHP experts can make suggests on how to improve the script it would be appreciated

Binfone Telecom Launches VoIP Website for Call Centers and Teleservices

Note: Usually I wouldn't find this interesting enough to post but half way down the press release I found a little tid-bit about Asterisk support. I have put it in italics below.

BinFone Telecom, a leading provider of VoIP for call centers, has launched www.binfone.com, a web site to serve the global call center and teleservice industries. The BinFone.com site offers news and information to call center and teleservice center operations and features articles discussing the reliability and cost savings of VoIP for this growing industry.

Snippet: BinFone offers a full product suite for call origination from the US including toll free numbers, and local numbers (DIDs) in all major markets in the US. The company offers both Inter-Asterisk Exchange (IAX and IAX2) termination and Session Initiation Protocol (SIP) termination. In addition, it supports Asterisk the open- source Linux PBX, and ViciDial, an Asterisk-based Predictive Dialer.

Click Here for Full Press Release

From Analog to VoIP: Asterisk Brings Telephony Together Under One Open-Source Platform

Note: I found this interview today with Mark Spencer creator of the Asterisk Platform.



Six years ago, Mark Spencer started his own Linux technical support business. Unlike other tech startups at the time, he spent his money frugally. Spencer had to; he didn't even have enough to pay for an office PBX system, which can cost up to several thousands of dollars.

"I had about $4,000 to start it out with, and I wasn't about to buy a phone system, so I figured I'd just make one," Spencer says.

He created Asterisk, a software platform PBX system, and open-sourced the code in 1999. Asterisk was not particularly useful to others outside of Spencer's own needs for his company, until a few years later when community contributions added support for more industry-standard telephony hardware, and modern Internet voice communications technologies, like Voice-over-IP (VoIP), to succeeding versions.

Click Here for the Full Interview with Mark Spencer

VoIP gear highlighted at CES

Note: I love seeing VoIP more and more getting media attention. The more this happens the more people will learn of Asterisk.

Phone calls over the Internet, made using voice over Internet protocol (VoIP) technology, was popular at the International Consumer Electronics Show (CES) in Las Vegas this year, with a number of companies showing off handsets and other peripherals to make it easier for the average user to connect to low-cost or free voice calls.

Many of the products were new handsets that connect to laptops or PCs via a universal serial bus (USB) plug, although a few companies showed off wireless handsets able to make calls directly over Wi-Fi signals, freeing users from their computers.

YapperNut delivered one of the more unique USB handsets: a computer mouse that doubles as a phone.

"People might forget to bring some peripherals with them on trips, but nobody forgets a mouse," chief marketing officer at YapperNut, Harrison Tang, said. The company is banking on growth in VOIP to raise interest in the mouse, dubbed YapperMouse, which vibrates and rings when calls come in, can hold your favorite ringtone, and only needs to be picked up to answer a call.

Click Here For Full Article

Announce: Enchance Me 1.004 Released!

Today I released Enhance Me 1.004 for AMP 1.10 and Asterisk @ Home 2.2.

These utilities allow for speed dialing, a revised version which uses AMP to store speed dial numbers and NAMES so that only operators of the web interface can add and delete speed dials. Data is stored in mysql instead of Asterisks’ DB. This stops the waste of the 300 series extension numbers.

Instructions on how to disable A@H's speed dial and insert my code is included.

Additionally, paging utilities for paging of all phones is accomplished with customized meetme routines.

Intercoms between stations are also documented with code. This code uses 0 + extension number to intercom the phone. This stops the waste of a separate ext number for intercoms and avoids the outside user from reaching an intercom line by dialing it directly. (If you use auto answer and an outside user dials the auto answer line they can sit back and eavesdrop)

A Cisco XML interface into AMP’s user database is used to display user names and their extensions on the phone. This is useful for new / temporary employees. This routine avoids displaying the intercom extensions set up.

A text recommendation file that covers a few changes to A@H such as fixing the backup link on the maintenance page is also included.

Click Here For Download

Enjoy,
Paul Norris
paul@siliconvp.com

Nerd Vittles: Introducing TeleYapper: Free Asterisk Message Broadcasting System, Part II

Nerd Vittles today presents the second in our series of articles on HOW-TO build and deploy a free Message Broadcasting System with Asterisk.

"... for neighborhoods, schools, little leagues, fundraisers, municipal governments, and anyone else that just wants to pester folks with annoying, but free, prerecorded phone calls."

Today's installment gets the whole enchilada dialing and broadcasting messages to callees entered in TeleYapper's MySQL database.

Click Here for Full Article

Release: ruby-agi-1.0.2 released today!

Hello,

I am happy to announce the release of ruby-agi-1.0.2 This is a stable release of ruby-agi.

ruby-agi is available at
http://rubyforge.org/projects/ruby-agi/

You can also install ruby-agi via gem.
To install ruby-agi gem package, try
% gem install ruby-agi


Feel free to send me your feedback, feature request and bug report.

Thank you,
Mohammad Khan

News: More Asterisk Case Studies

Note: VoiceIP Solutions has posted some more Asterisk Case Studies. This time adding information for a Mortgage Office and Call Center.

Case Study: Call Center

A client that wanted to setup a 50-user office including a 20-user call center, asked VoiceIP Solutions to craft a solution that would leverage Linux and the Asterisk Open Source PBX to reduce their startup costs and allow for growth.

Click Here for the Full Case Study

Case Study: Mortgage Company(10 Wireless Users)

A mortgage company with 10 employees needed a low-cost telephone system for their growing business. VoiceIP Solutions was able to provide a low-cost PBX based on the Asterisk Open Source PBX platform. This free software runs on the Linux operating system and utilized standards based server hardware to keep costs down.

Click Here for Full Case Study

Click Here Here For All Case Studies

News: Dual GSM - VoIP Phone Announced

Note: The fine people over at VoIP News Net have post info on a new GSM VoIP Phone announced at CES in Las Vegas.

"Another VoIP related announcement from the Consumer Electronics Show in Las Vegas:

Telephony company UTStarcom announced the GF200, a dual mode GSM and WiFi VoIP phone. An exact launch date or price hasn't been announced yet, but it's expected to come out in the second quarter of 2006.

The phone will allow users to make calls using GSM cellular networks or 802.11b/g WiFi VoIP connections. The phone may come out in Europe soon, but deployment in the United States is being delayed because sources say that the company is having trouble finding any major US carriers to support the phone."

Click Here for the Full Article

Click Here for More Information about UTStarcom

Announce: New Freelance Site for Asterisk Consultants

Note: Steve Totaro has posted info about his new website to help people find Asterisk Consultants. It is in the Beta stage right now and will be for the next 6 monthes according to Steve.

Hello all,

I have created a beta site for "Asterisk Gurus" or Consultants to bid on projects posted by customers needing to have work done. It is very similar to scriptlance or any of those other sites but it is dedicated to Asterisk and related issues so hopefully only really qualified Asterisk consultants will bid on your projects. If you post at one of those other sites, you wind up with 99% of the people who bid unable to complete the project and they waste your valuable time.

Asterisk is a very specialized skill and with our rating system, we can quickly identify who the good "Asterisk Gurus" are and not waste time with the wannabes.

This also seems to be a very good replacement for the "Bounty" system on www.voip-info.org <http://www.voip-info.org/> . I am sure we can figure out how to split costs owed to the "Asterisk Guru" between customers.

It is VERY beta right now but I think it is also fully functional. Any reference to payments, deposits, $$$, etc can be ignored. The service is free for now and will stay that way for at least the next six months.

However, I am may add a PayPal donation link since this is certainly not free for me. Of course there is no obligation to donate but I would appreciate it. Heck, if there are enough donations then the site could remain free permanently.

There are some small issues since the script is wrapped in another script, but I am aware of this and will find a fix shortly. Besides that, I could use any input on usability, additions, categories not listed, or whatever jumps to mind.

I will also be adding a section to post resumes and other permanent job postings.

Please test it out and let me know what you think.

http://www.asteriskhelpdesk.com

Thanks,
Steve Totaro

News: Switchvox Named to Digium Premier Partner Group

SAN DIEGO--(BUSINESS WIRE)--Jan. 4, 2006--Switchvox, a leading provider of IP PBX phone systems for SMBs, today announced it has been named a premier participant in the Digium Partner Program. Digium® Inc., the creator of Asterisk® and pioneer of open source telephony, developed its Partner Program in response to the growing ecosystem of Asterisk and the demand for open source telephony platforms.

The goal of the partnership is to form a closer relationship between Digium and Switchvox, which implemented the Asterisk telecommunications platform in its flagship IP PBX system for SMBs. Code for Asterisk, originally written by Mark Spencer, CEO of Digium Inc. and contributed to by over 350 individuals, has been used by open source software engineers around the world to develop leading-edge telephony applications.

Click Here for the Full Article

News: Still an open Seat in London for Next Weeks Signate intro to Asterisk Course

Note: Paul Mahler has post info about an available spot at a London based Asterisk Course.

We still have a seat open in our Asterisk training course next week in London. You can find more information at our Web site: www.signate.com

I'm going to be teaching the class.

Paul

Announce: QueueMetrics 1.0 released!!

It is with great pride that we announce the availability of QueueMetrics version 1.0.

QueueMetrics 1.0 adds new error-handling features that are meant to make your user experience as smooth as possible. It also adds a JDBC testing module that spots common setup problems and helps finding them. The realtime engine was modified for better responsiveness and the memory usage of the system was improved, in order to run even larger analyses.

QueueMetrics 1.0 allows data storage on both flat files and MySQL databases for bigger call centers. And of course comes with a 64-page user manual that covers all aspects of it.

So it is time to upgrade your systems to version 1.0. And would like your feedback too: what would you like to be in the next releases? We plan to vastly improve the realtime panel as the number one priority. But you use QueueMetrics every day, so we would like to hear from you on how the product can be improved.

The latest version of QM can be download from: http://queuemetrics.loway.it/download.jsp
Click Here for More Information

Yours,
lorenzo emilitri
Loway Research

Announce: Sacramento Asterisk Users Group

At our new location 7-9pm Janurary 12th. No fee to attend
Exit Certified
8950 Cal Center Drive
Suite 110, Bldg. 1
Sacramento, California, USA
95826

http://www.sacaug.org/
(map links on the sacaug webpage)

We will be doing an install of astlinux with simple configuration as well as discussing upcoming contests for which we have prizes donated by Digium and TheVoipConnection.com it should be fun :)


--
Trixter http://www.0xdecafbad.com
Bret McDanel

Announce: Live Demo of DRUID Asterisk Management Interface

Hi,

We have recently setup a Live Demo of DRUID our Asterisk management interface product. Also I'd like to thank all of you that took the time to download the trial edition and give us your feedback. WE've tried to incorporate as much of that feedback into our new updated release.

Feel free to download the trial, checkout the live demo or buy a copy :)

http://www.voiceroute.net/site/demotrial.php
http://www.voiceroute.net/site/index.php
http://www.voiceroute.net/site/faq.php

Enjoy

--
regards
Vikram

Announce: SoCal Users Group Meeting Schedule

The SoCal Asterisk Users Group will be meeting at the Heritage Park Public Library on the corner of Walnut and Yale in Irvine on the 3rd Thursday every month. The following dates are already secured:

Thurs Jan 19
Thurs Feb 16
Thurs Mar 17

Irvine Heritage Park Library
(949) 936-4040
14361 Yale Ave
Irvine, CA 92604
Google Directions: http://tinyurl.com/9vq3e

Release: AstLinux 0.3.5 is out now

Hello everyone,

I have placed AstLinux 0.3.5 in the -testing tree. It has the following improvements:

* Updated e2fsprogs
* /etc/rc fixes to improve filesystem checking
* MSMTP fixes (should work now)
* openvpn init support

Also, I have added support for passing module parameters in /etc/rc.modules. I would also like to support passing module parameters in the ZAPMODS rc.conf variable. I'll be working on this...

Existing users can upgrade by running "astup testing".

Please test! Thanks!

--

Kristian Kielhofner

News: Uniden announces two new VoIP phones

niden unveiled two new VoIP-enabled phones at the Consumer Electronics Show in Las Vegas: the Uniden WIN 1200 and the Uniden UIP165P.

The first, the Uniden WIN 1200 a 5.8GHz digital cordless phone supports both traditional landline and VoIP phone service. VoIP service is handled by Microsoft's Windows Live Messenger, with MCI Web Calling Service providing net-to-landline service.

"This alliance (with Microsoft) will unlock the world of VoIP communications to a broader demographic of users," said Blake Irving, corporate vice president of the MSN Communication Services and Member Platform group at Microsoft. "Rounding out software and services with great devices is a key pillar of Windows Live, and the Uniden WIN 1200 and Windows Live Messenger makes advanced voice technology feel familiar." The Uniden WIN 1200 has a list price of $99.99.

Click Here for More Information

Announce: New AMPortal and Asterisk Debs

Hi folks

You are welcome to try our (Xorcom)'s latest debs (for Xorcom Rapid, or Debian Sarge in general)

"Unstable": Asterisk and AMPortal:

The repository is available at:

deb http://rapid.dotsrc.org/ unstable/
#deb-src http://rapid.dotsrc.org/ unstable/


The commands you are looking for are:

apt-get update
apt-get dist-upgrade
apt-get install amportal


"Experimental": Asterisk 1.2:
At the moment they are not that experimental anymore and should be ready for use, but are not well-tested yet.

To use it, define both sources:

deb http://rapid.dotsrc.org/ experimental/
deb http://rapid.dotsrc.org/ unstable/
#deb-src http://rapid.dotsrc.org/ experimental/
#deb-src http://rapid.dotsrc.org/ unstable/

and use the usual:

apt-get update
apt-get dist-upgrade
apt-get install rapid-scripts

amportal should also work there, but has not been tested at all.

--
Tzafrir Cohen
http://tzafrir.org.il
tzafrir@cohens.org.il
ICQ# 16849755

Release: ---AEL2--- Try it out!

Hello--

I've just written and submitted a new module for asterisk, to the
asterisk bug database.

See: http://bugs.digium.com/view.php?id=6021

There is a file there you can download, AEL2v0.3.patch.bz2

and I created a wiki page: http://www.voip-info.org/wiki/view/Asterisk+AEL2

Why did I do it? Because I was very impressed with AEL, but the current AEL compiler isn't real good at pointing out problems. Mostly, it seems to silently ignore problems.

But I have a better idea. I want the compiler to check the "living daylights" out of the dialplan code. I want subtle errors that otherwise would trip up asterisk and hang up a call to be found at load time, not run-time! I remember all the classes I took long ago, that kept pounding away at the fact that finding and fixing errors at later stages are exponentially more expensive than finding and fixing them early in the design cycle... and I think the same applies to dialplans.

So I wrote AEL2 from the ground up. No newline dependencies, as free-form as C. After parsing, which will reveal syntax errors, a second pass is made hunting for semantic errors, like misspelled applications, bad expressions, etc. I'll tack on a list of the differences and checks at the end of this message. Code is generated in a third phase, if no errors are found in the input.

News: Second edition of my *(Asterisk) book has been released

The second edition of my Asterisk book "VoIP Telephony with Asterisk" is now in print. It's reorganized and expanded.

TKS

Paul Mahler
pmahler@signate.com

www.signate.com

Announce: Web MeetMe Seeking Beta Tester

I have V2.0.0 about read to roll. There's quite a few new features, clean-ups and
bug fixes. I hope this will be the last major update for awhile, and as such I'd like to squash any issues before releasing it.

Click Here to Download Web MeetMe v2.0.0

-New
** Dynamic day/date listboxes to prevent invalid date entries
** Textbox focus and tabbing order
** CDR-like view at past conferences to show participants
** Vastly simplified interface to 'out call' and invite a participant
** Optional branding: Changing titles and logos based on a define

-Fixed
** The Previous link on the Current/Future views reverted to the Past view
** A number of titles better reflect the intent of the page they are on

Please contact me off list if you'd like to give it a try.

Thanks,
Dan

News: RED ALERT! Bug marshals need your help! NOW!

In the developer community,

We now have over 250 open issues in the issue tracker. We need all the help we can get to move through this and clear it out so it's manageable again.

We need your help with

* Testing
* Code reviews
* Portability Issues

All confirmations of a test makes it easier to make a decision to commit. A lot of the issue reports have no test results at all, and it's hard for us bug marshals to set up and test all possible solutions and configurations.

Please, please, dedicate some time to the issue tracker. Log in, pick a report and test, test, test, test.

Thank you for your help in making progress!

http://bugs.digium.com/main_page.php

/Olle

Nerd Vittles: Introducing TeleYapper: Free Asterisk Message Broadcasting System

Nerd Vittles today presents the first in a series of articles on HOW-TO build and deploy a free Message Broadcasting System with Asterisk.

"... for neighborhoods, schools, little leagues, fundraisers, municipal governments, and anyone else that just wants to pester folks with annoying, but free, prerecorded phone calls."

Introducing TeleYapper: Free Asterisk Message Broadcasting System

http://mundy.org/blog/index.php?p=95

Announce: New Manager Client Program

Note: Bill Michaelson has posted details about a new Asterisk client program.

Here is a work-in-progress that provides pop-up note-taking windows based on caller-ID, outgoing call dialing from directory lookup selection, and other stuff.

I hope it's useful to folks.

Link:
http://asteroid.from.net

Announce: chan_capi-cm 0.6.2 released

Hi all,

I just released version 0.6.2 of chan_capi-cm for Asterisk.

The package and notes can be found at sourceforge:

http://sourceforge.net/projects/chan-capi

Armin

PS: This version of chan_capi-cm is already part of OpenPBX trunk.

Rumor Report: Allison on Free 411

Note: Joe Pukepail has posted the following rumor reguard our Asterisk voice. Decide for yourself.


I heard on the radio about 1-800-FREE411 and tried it out, I was very suprised to hear allisons' voice for the digits. Not sure if they are using asterisk for the backend on this or not.

Try it out its Free!
http://www.snopes.com/inboxer/nothing/
free411.asp

(not afflicated with it in any way).