Asterisk-Java 0.2-rc1 released

Asterisk-Java 0.2-rc1, 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 FastAG protocol and the Manager API.

The 0.2-rc1 release candidate focuses on the new features of Asterisk 1.2-beta1 though it still supports Asterisk 1.0.x.

The changes include:
* Support for the new Actions, Events and Commands of Asterisk 1.2
* New support for event generating Actions, i.e. Actions that send their result as a series of Event rather than the usual ManagerResults. See the sendEventGeneratingAction() methods in ManagerConnection for more information.
* New base class for AGI scripts that allows you write cleaner AGI scripts as you don't have to pass the channel variable to all methods.
* New convenience constructors for manager actions
* Some minor bug fixes

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
http://asterisk-java.sourceforge.net

astGUIclient version released 1.1.6

We've released another update to our Asterisk GUI Client suite: 1.1.6

http://astguiclient.sf.net/

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 the VICIDIAL web-client, added compatibility with the Asterisk 1.2 release tree, streamlined several server-side apps and added cpu percentages to our stats logging scripts.

As of this release, all client apps and daily administration functions can be access through a web browser and we have tested our new AJAX-enabled(PHP, Javascript and XMLHTTPRequest) VICIDIAL client in production with great results.

Let me know what you think.

Thanks,

MATT---
http://astguiclient.blogspot.com

Web-MeetMe v1.3.3 Released

This came through on the Asterisk User list today.


Work intrudes again and I will not be able to get to modifying the db and gui
to support per-conference flags as soon as I expected. So I have released an update with what I do have available.

[Location]
http://www.fitawi.com/Asterisk

[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
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 (but outside of my interest to do so (patches welcome))
7. Users can only monitor, update or delete their conferences
8. Verified administrators can monitor, update or delete any
conferences.
9. Updated to CVS-Head (a couple weeks ago, will target 1.2 soon)
a. Changes to the Manager interface may have caused support for 1.0.X to slip, I cannot test that)

There is one functional issue to be addressed, and that is that MeetMe tracks conference participants by channel. From a conference management perspective it makes more sense to track the participant by caller-id. I have a patch for 1.0.X on my site, but have not polished one for CVS-Head or the 1.2.0beta release.

Thanks and enjoy,
Dan

Asterisk 1.2.0-beta1 Released

The first beta of Asterisk 1.2.0 has been released! It is available from the ftp.digium.com FTP servers, as well as the Digium CVS servers (under the 'v1-2-0-beta1' tag).

This version of Asterisk represents a significant improvement in features, stability and compatibility over the 1.0.x releases. Some of the major new (or upgraded) features include:

* Asterisk Realtime Architecture
* Asterisk Manager Interface
* Asterisk Extension Language
* Dialplan functions
* More powerful dialplan expression parser
* Portability enhancements for FreeBSD, OpenBSD, Solaris and Mac OS X
* ... and many more!

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!

PhoneCALL version 1.0 Administrative Manual -Released

Greetings Everyone!

The version 1.0 of the PhoneCALL Administrative Manual has been released. It is more of an outline of the features and interface, and we'll be adding lots of more detailed information in the manual over the next few days/weeks.

Of course, we'd love to get your input on the manual and areas we need to clarify or even some new sections in the manual that would help explain PhoneCALL and how it works.

You can find the PDF version of the manual in the Downloads, or you can view the HTML version here:
http://www.vecsector.com/phonecall/demo/manual

Enjoy!

Dustin Wildes
VecSector, LLC

HooDaHek 0.4 Released

HooDaHek, the caller ID and instant messaging notification service for Asterisk boxen, is now updated to version 0.4.

Information/download here:
http://www.nathanpralle.com/software/hoodahek.html

Changes:
- Changed the AIM bot to use Net::OSCAR instead of Net::AIMTOC since AOL managed to break TOC in some way. Tired of such shenanigans, so switched. Reminder to self: Call Apple.
- Implemented HiRes timing for the Bot so it pops up CallerID information tons faster than it used to -- often by the first ring.

Enjoy.

Nathan

Register Today for Fall 2005 VON: "TheDestination for IP Communications"

I found this email from Jeff about VON 2005:


While flying to London yesterday, I spent some time thinking about VON and how while some things change, other things about VON remain the same.

Since our first VON event in the Spring of 1997, our VON events have over time become the worldwide Destination event for IP Communications.

In fact, while we are actively marketing Fall 2005 VON using various channels around the United States, it is the continued strong word-of-mouth buzz that is bringing in delegates from around the world. So far, there are delegates registered from 40+ countries including: Argentina, Aruba, Australia, Austria, Belgium, Brazil, Canada, Chile, China, Costa Rica, Denmark, Dominican Republic, Finland, France, Germany, Ghana, Hong Kong, Hungary, India, Ireland, Israel, Italy, Japan, Korea, Singapore, Slovenia, South Africa, Spain, Sweden, Switzerland, Taiwan, Turkey, UK, UAE, USA and Uzbekistan.

I expect the buzz to be pretty strong when the doors open in less than four weeks. The 330+ exhibitors in our "Sold Out" exhibit hall represent our largest exhibit hall...ever! (and has grown by more than 100 exhibitors since Spring 2005 VON.)

The Fall 2005 VON conference sessions are returning to the size we experienced five and six years ago.

The registered delegates in Boston are all part of the ecosystem that makes up our VON events. There will be people representing just about all aspects of the IP Communications food chain.

Note: Vendors who are interested in exhibiting at Spring 2006 VON should consider signing up now. The pulvermedia Sales team is projecting that the exhibit hall at Spring 2006 VON will be close to sold-out before we arrive in Boston for the commencement of Fall 2005 VON.

Experience the Journey and register today for Fall 2005 VON, "The Destination for IP Communications."
Please visit: <https://secure.pulver.com/von/register.html> to register.

Best regards,

Jeff

Release: *starShop: Open Source Calling Shop Monitoring System

I am following up on this story that was broke on SineApps. This app based on the Asterisk Platform is used to keeping track of billing of phone calls. Here is an excerp from there website:


" *starShop is a professional and powerful billing, monitoring and management system for Calling Shops, Internet cafes, libraries, schools, hotels and any hospitality place that provide phone calls for public use."


Click Here for more Information

Release: PhoneCALL v2.6.1 - Released

Hello All!

Just a notice that our PHP/Smarty-based GPL version of PhoneCALL version 2.6.1 has been released, and is the current stable release.

http://www.vecsector.com/phonecall

We're always looking for feedback/testers to help us enhance it and make it even easier for everyone to use. The current version is designed around the advanced Asterisk user, and we are working on a more 'restrictive' model for different types of users in the system, for example:

1) User-based logins so users can control their phone options (like DND, Call CellPhone, Text Message) or update their name, email
2) Admin-based logins that control the general 'call flow' - but not administer any of the scripts/macros and can only see the information for the tenant they are assigned.
3) Site-Admin has full access to all accounts/scripts, etc... like root account (current setup)

We're taking feature requests, and all feedback is welcome.
Thanks!



Dustin Wildes
VecSector, LLC

HooDaHek 0.3 Released

HooDaHek 0.3, the database handler and IM notification bot software for Asterisk, has been released.

http://www.nathanpralle.com/software/hoodahek.html

Nothing terribly fancy in this release except a few bugfixes and the bot now recognizes a few commands to search the database.

Nathan

New Beta IAX Statistics Program

Found this on the User list from the SineApps boys:

Hot off the wire:

http://www.sineapps.com/news.php?rssid=927

Hi, we have put together a small application for Windows to allow you to check IAX network statistics.

Basically all you need is the .Net framework and the
user/pass/host/extension/context details.

There is one parameter available when you start. This is dial string. It is made up as follows:

user:pass@host/extension@context

You can download it from here: (415K)

http://www.sineapps.com/SineStatsIAX%20Installer.zip

Please let me know how you go with it.

--
Cheers,

Matt Riddell

New Digium 2-port PCI Hardware

Digium has released a new 2-port interface card. Here is the infomation from there site:



The TE210P is the next generation of Digium hardware that improves performance and scalability through bus mastering architecture. The TE210P supports E1, T1, and J1 environments and is selectable on a per-card or per-port basis. This feature enables signaling translation between E1, T1, J1 equipment and allows inexpensive T1/J1 channel banks to connect with E1 circuits. Because the TE210P improves I/O speed by up to 10 times, the result is reduced CPU usage and increased card density per server.

Digium has designed the TE210P to be fully compatible with existing software applications and it is fully integrated with the Asterisk Open Source PBX/IVR platform. Also, the open source driver supports an API interface for custom application development. With the combination of Digium Hardware and Asterisk software, numerous combinations of telephony configurations are possible. From the traditional PBX to VoIP Gateways, Digium solutions are paving the way for a new generation of worldwide communications.

The TE210P supports industry standard telephony and data protocols, including Primary Rate ISDN (both N. American and Standard Euro) protocol families for voice, PPP, Cisco, HDLC, and Frame Relay data modes. Both line-side and trunk-side interfaces are supported, also included are advanced call features.

The TE210P supports a 3.3v PCI slot only - typically available on newer motherboards and in 64-bit PCI bus architectures. The TE205P is for use only with a 5.0 volt PCI slot.

Click Here for Full Specs

New astGUIclient version released 1.1.5

The great people that work on AstGUI have released another update to there Asterisk GUI Client suite: 1.1.5

http://astguiclient.sf.net/

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 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 refined the web-client, added a few features to it and we also include an alpha version of the VICIDIAL web client. On the server side, we have moved server config variables to the web admin section for ease of modification, especially in multi-server environments.

Let me know what you think.

Thanks,

MATT
http://astguiclient.blogspot.com

FCC Issues Rule Allowing FBI to Dictate Wiretap-Friendly Design for Internet Services

I found this in my email box. I will be watching how this pans out. The EFF posted this story on there website. Here is a excerpt:

"Tech Mandates Force Companies to Build Backdoors into Broadband, VoIP

Washington, DC - Today the Federal Communications Commission (FCC) issued a release announcing its new rule expanding the reach of the Communications Assistance to Law Enforcement Act (CALEA). The ruling is a reinterpretation of the scope of CALEA and will force Internet broadband providers and certain Voice-over-IP (VoIP) providers to build backdoors into their networks that make it easier for law enforcement to wiretap them. The Electronic Frontier Foundation (EFF) has argued against this expansion of CALEA in several rounds of comments to the FCC on its proposed rule.

CALEA, a law passed in the early 1990s, mandated that all telephone providers build tappability into their networks, but expressly ruled out information services like broadband. Under the new ruling from the FCC, this tappability now extends to Internet broadband providers as well."

IAX Phone Pro Beta - New Version Available

[ New Beta Version - Beta Extended ]

A new version of IAX Phone Pro Beta is available. A few bugs have been fixed and the beta has been extended until October 12, 2005 (the date of AstriCon 2005). You can download either a new install (be sure to un-install the old version) or just a new binary.

[ IAX Phone Pro Features ]

* Dial/answer/hold/recall/reject
* Multi-number advanced speed dial.
* Standard and innovative "Tool Bar" skins.
* Handles "iax:", "sip:" and "tel:" URLs
* Integrated web browser for "co-browsing"
* Integrated call recording and playback.
* Advanced phone book with CSV import.
* Advanced call log with CSV export.
* Speaker Phone
* Audio mute.
* Auto answer.
* Intercom calling with password.
* Multi-server registration.
* Audio Codecs: uLaw, aLaw, GSM, iLBC, Speex.
* Server-by-server codec setting.
* Call statistics.
* Local or server-side call forwarding.
* Local or server-side do-not-disturb.
* TAPI integration for Outlook, ACT, Goldmine, etc.
* Direct IP to IP calling
* Dial by IAX or SIP URI (URL)

[ Try Out Phone URI/URL Dialing ]

IAX Phone Pro supports the ability to handle telephony URIs (links). This feature is great for call centers or web-based contact management solutions. When you install the phone, it configures your copy of Windows to pass all links marked as "iax:", "sip:", or "tel:" to IAX Phone Pro. IAX Phone then does its best to place a call to the destination number.

You can create these links by using the iax, sip and tel URI schemes. Simply use the following examples as a guide:

Call Ipsando HQ
Call Extension 1000
800 Directory Information (US Only)
Olle Johansson over SIP (requires the
SIP-Over-IAX)

IAX and SIP accept IAX or SIP URIs respectively. TEL allows you to enter any extension or dialable number. Note that your browser /may/ ask you to authorize each of the URI types (iax, sip, and tel) the first time you click on them. You must select OK in order for the calls to go through.

[ Download IAX Phone Pro ]

https://www.astricon.net/phone/ipbeta.php

Steven Sokol
CEO/Manager
Sokol & Associates, LLC

OPAL now supports IAX2

Craig Southeren announced today that OPAL (http://www.voxgratia.org) now provides support for the IAX2 protocol(Written by Derek Smithies and released under the MPL). This support allows you to use chan_woomera (http://www.pbxfreeware.org) driver developed by Anthony Minessale II to interconnect your asterisk systems and use the IAX2, SIP, and H.323 protocols.

I would like to thank everyone involved in Cluecon for all their support!

Thanks guys!

Brian West
Asterlink.com

AstriCon 2005 - Early Bird Registration Open (Free IAXy To First 50!)

// AstriCon 2005 - Oct 11 - 14, 2005 - Anaheim, California USA //



[ REGISTRATION NOW OPEN]
------------------------------------------------------------------
Digium and Ipsando are pleased to announce that AstriCon 2005 Early Bird Registration is now open. Early Bird registration can save you 20%($110.00 USD) off the full conference admission. The first 50 to qualify for Early Bird by purchasing an AstriCon "All Access" Pass will also RECEIVE A FREE IAXy from Digium*.

Register Now:

[ WHAT IS ASTRICON? ]
------------------------------------------------------------------
The only conference dedicated exclusively to Asterisk. AstriCon includes:

* Two Pre-Conference Events:

- The Asterisk Developer Summit
- Meet Asterisk! - An Introductory Seminar

* A full day of Asterisk Tutorials:

- Beginner: Learn to install and implement Asterisk
- Intermediate: Learn tips and tricks for enhancing your PBX
- Advanced: Scale and cluster Asterisk, improve security

* Two Full Days of Conference:

- Keynote from Asterisk creator Mark Spencer
- Presentations from lead developers
- Asterisk Industry Perspectives
- Panel discussions & round tables
- BOF Sessions

* The Asterisk Exposition & Trade Show:

- Service Providers from around the globe
- IP Phone manufacturers and distributors
- VARs and Integrators
- Training & Support Organizations

[ WHEN & WHERE IS IT?]
------------------------------------------------------------------
AstriCon 2005 will be held from October 11 through October 14 at the Hyatt Regency Orange County in Anaheim California.

[ WHO WILL BE THERE? ]
------------------------------------------------------------------
Last year's AstriCon drew nearly FIVE HUNDRED attendees. The goal for this year is nothing short of doubling the previous attendance. Attendees include: enterprise users, Internet telephony service providers, competitive local exchange carriers, interconnect vendors, consultants, systems integrators, VARs, developers, ISPs, and hobbyists.

[ EXHIBIT or SPEAK at ASTRICON ]
------------------------------------------------------------------
For information on speaking opportunities or for exhibition information, contact us.

Email: info@astricon.net
Phone: +1 816 256 8916
IAX2: IAX2/guest@pbx.sokol-associates.com

See you at AstriCon!


* Winners will be able to pick up their IAXys from Digium at the Digium
booth at AstriCon.

AstLinux 0.2.8 released

I have just finished up work on a new release of AstLinux, the
embedded/live cd/minimal built from scratch Linux distro centered around Asterisk.

Most of the work for 0.2.8 has been on the ISO image (live/install cd), the init system and further system customization with new rc.conf variables.

- The ISO image now allows you to install AstLinux to a local disk/CF card, write images from Windows, and run AstLinux (from RAM) without having to touch the hard disk (live cd). It now auto-detects your cd drive and runs from RAM, so you can change your password, etc. It even includes a nifty Windows autorun feature, all in about 45 mb (small enough for business card CD's)!

- The Windows install package has now been updated to use the same menu as the CD. Also, the AstLinux disk images are now stored compressed, so the installer will only use about 27mb of your disk space in Windows!

- AstLinux has seen dozens of changes and feature improvements. Far too many to list here, but for the full changelog please visit:

http://www.krisk.org/files/astlinux-i586/etc/astup.note

Existing AstLinux users can simply type "astup" to be upgraded to the newest release, while existing and new users can use the disk images available at:

http://www.astlinux.org

Thank you, and I look forward to meeting some of you at ClueCon over the next couple of days!

P.S. - 0.2.8 still only occupies around 26mb of disk space :).

--
Kristian Kielhofner

IPSwitchBoard version 0.124 Released

Thorben Jensen has posted details of the latest release of IPSwitchBoard:

Version 0.124 - 27 July 2005

Improved Billing facilities:

-Rate can be specified with four decimals
-Connection Fee can be supressed by specifying a negative number
-Rates can be defined for a time period (ex.: 18:00 - 23:59)
-Description can be added for a Call Rate

Download Here

NEWS RELEASE: Sangoma Hardware Selected By Pandora Networks As An Integral Part Of On Demand IP Communications Solution

Sangoma Technologies Continues To Extend Leadership Position In Managed VoIP Networking Space

Toronto, ON - August 2, 2005 - Sangoma Technologies Corporation (TSXV: STC) (www.sangoma.com), a leading provider of connectivity hardware and software products for VoIP, TDM voice, WANs and Internet infrastructure, announced Pandora Networks (www.pandoranetworks.com) has selected Sangoma's AFT Series of TDM hardware as an integral part of its On Demand IP communications solution. Based in Emeryville, California, and established in 2001, Pandora Networks has developed WorkSmartTM, an On Demand IP communications solution to improve business communications.

Capable of handling both voice and data and supporting all popular open source projects, Sangoma's AFT Cards offer a new market standard for performance, reliability, compatibility, support, and ease of installation. With less demand on the host CPU, drivers take advantage of the AFT technology to substantially reduce the processing required to handle TDM voice calls. This reduces the CPU's workload and results in fewer dropped calls, less jitter and better voice quality for callers.

AFT card features:

Compatible and Flexible - Sangoma's voice/data cards are self-sensing for 3.3v and 5v PCI slots and software configurable for T1, E1 or J1. They share interrupts properly between themselves and other PCI compatible devices, supporting unlimited numbers of cards per PC chassis. Conforming to the 2U form factor, both in height and length, the AFT cards allow users to install many cards in a slimline 2U chassis to maximize server capacity.

High performance - Sangoma cards have been carefully designed to reduce CPU loads in TDM environments, improving system performance and reliability on larger systems.

Trust and quality - As part of its commitment to quality assurance, each Sangoma card is individually inspected and burned-in prior to shipment in protective anti-static wrap complete with cables, manuals and CDs. The result is an almost zero dead on-arrival rate and high reliability in service.

Unparalleled support - Sangoma's fully engaged engineering support for both its hardware and software is unrivalled in the industry