The Server move is complete

Yay!

All sites are moved to the new server and I am a very happy geek. Have fun with the increased speed everyone.

I'll be busy for a while updating and setting up various things as always, but i'll try and post as much as i can.

nite.

Digium Announces the Launch of Asterisk 1.2

Digium Announces the Launch of Asterisk 1.2
Asterisk 1.2 has over 3,000 improvements, upgrades, fixes, and additions

Digium Inc., the creator of Asterisk® and pioneer of open source telephony, announced Asterisk 1.2, today at the IP.4.IT conference in Las Vegas, Nevada. Asterisk 1.2 is the first major revision to Asterisk since the release of Asterisk 1.0 in September 2004, and includes over 3,000 feature additions and improvements to the overall performance and efficiency of memory usage. Asterisk, the world’s first open source PBX, offers a strategic, highly cost-effective approach to voice and data transport over TDM, IP and other architectures.

"We have been working very hard with the support of the Asterisk community to release version 1.2 of Asterisk," said Mark Spencer, president of Digium and creator of Asterisk. "As Asterisk plays an ever expanding role in the telecommunications industry, it's important to support the rapid development model of open source software – quickly moving features from concept to product while retaining software quality and architectural integrity."

A significant number of changes have been made to the core of Asterisk including code formatting, simplification and documentation. The Asterisk developer community extends all over the world, and the new changes incorporated in Asterisk 1.2 make it easier for new developers to get involved. New features include:

- A number of significant changes to the core to improve performance and memory usage
- Improved voicemail features
- Addition of the DUNDi (Distributed Universal Number Discovery) protocol
- Easier Asterisk configuration
- Creation of a Realtime Database Configuration Storage Engine
- More power added to the Asterisk Dialplan
- Introduction of Asterisk Extension Logic, a new, flexible method for configuring the dialplan
- New interface for dynamic IVR flow control
- Configurable access to general call features
- Improved SIP protocol support
- New features for the IAX (Inter-Asterisk eXchange) protocol
- Use of sound files for native music-on-hold
- Customized CDR Support
- PRI support improvements

Availability

Asterisk 1.2 will be available for download from the Asterisk website,FTP and CVS servers after 5:00PM Pacific Standard Time on November 16th.
About Asterisk

Code for Asterisk, originally written by Mark Spencer of Digium Inc., has been contributed from open source software engineers around the world. It supports a wide range of TDM protocols for the handling and transmission of voice over traditional telephony interfaces. It also supports US and European standard signaling types used in standard business phone systems, allowing it to bridge between next-generation voice-data integrated networks and existing infrastructure. Using the Inter-Asterisk eXchange (IAX™) Voice over IP protocol, Asterisk merges voice and data traffic seamlessly across disparate networks. While using packet voice, it is possible to send data such as URL addresses and images in-line with voice traffic, allowing advanced integration of information.

The Digium logo, Digium, Asterisk, and the Asterisk logo are trademarks of Digium Inc. All other trademarks are property of their respected owners.
About Digium

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

Digium solutions reduce the costs of traditional TDM and VoIP implementations through Open Source, standards-based software and next-generation gateways, media servers, and application servers. Digium hardware supports traditional voice protocols, including PRI, RBS, FXS, FXO, E&M, Feature Group D, Groundstart, Loopstart, and GR-303. Data protocols include PPP, Cisco HDLC, and Frame Relay. For packet voice, Asterisk supports IAX™ (Inter-Asterisk Exchange), SIP, MGCP, Cisco Skinny® (SCCP), and H.323 VoIP protocols.

[Asterisk-Users] DeStar 0.1 released!

Hi everyone,

We are glad to announce to all the Asterisk community the first
release of DeStar[1], a web based interface to manage the Asterisk
PBX.

DeStar provides high-level abstraction above the Asterisk
configuration, making it real easy to quickly setup a basic PBX, but
simultaneously allowing great flexibility for those out there
intending to manage medium-complexity Asterisk based telephony
systems.

DeStar main features include:
- Extensions management: SIP, IAX, Zap, and more.
- Auto-attendants support.
- Trunks management: SIP, IAX, Zap, ZapPRI, and more.
- Use of dialout patterns (i.e. local, national, mobile-phones, toll-free numbers, etc).
- Asternic Flash Operator Panel [2] integration.
- Call Detail Records search and graphical reports.
- Many application applets incluided: Voice Mail, Meeting Room, and more.

DeStar is written in Python and uses Quixote[3], Sqlite[4] and
Pychart[5]. You can download it from [1] or get it for the Debian
GNU/Linux testing and unstable distributions via apt.

A good starting point would be the Project Home Page[1] or the Project
Wiki[6].

You may suscribe to the destar users list entering [7], where ALL
questions are welcome.

For developers, the list suscription can be made in [8], ALL questions
are welcome too.

Or if you prefer, you may find us at the DeStar IRC channel, where
we'll be willing to answer your questions, discuss technical aspects
of DeStar or just attend your complains about it ;-):
Server: irc.freenode.net
Port: 6667
Channel: #destar

There's still a lot of work to do, so we encourage all of you who may
be seeking for alternatives to configure the Asterisk PBX to join us.
Testers, Programmers, Documentators, Translators, Graphic Designers,
Usability Analyzers and users in general are needed.

Thanks to all those who helped us to reach this point.

Now enjoy DeStar!

Best regards,

The DeStar Development Team.

---
List of links

[1] http://destar.berlios.de/
[2] http://www.asternic.org/
[3] http://www.mems-exchange.org/software/quixote/
[4] http://www.sqlite.org/
[5] http://home.gna.org/pychart/
[6] http://openfacts.berlios.de/index-en.phtml?title=DeStar
[7] http://lists.berlios.de/mailman/listinfo/destar-user
[8] http://lists.berlios.de/mailman/listinfo/destar-dev

[Asterisk-Users] Realtime Voice Changer Patch

I have just written a patch for Asterisk 1.2.0-rc1 that allows you to
install a voice changer on a channel.

http://www.lobstertech.com/voicechanger/

If you are a developer, please feel free to help add features and
clean up the patch so that we can hopefully get it in CVS some day.

- Justin Tunney

[Asterisk-biz] QueueMetrics 0.9.7 released

Hello list,

I am pleased to tell you that we have released a new version of QueueMetrics. This version focuses on MySQL support - in fact you can now move your queue_log data to MySQL and have QM analyze them. This offers a major advantage in terms of flexibility and performance for bigger call centers. MySQL support at this time is still experimental, but our test found no significant problems with it. We would love to hear from you if you find any bugs or would like to improve it.

We have also improved the real-time page for call center monitoring, showng a panel that tells you the status of your center at a glance, with no need to manually count how many calls are waiting or how many agents are free at the moment. We hope you'll like it.

To upgrade your current version of QM to 0.9.7, have a look at the upgrading.txt file. If you plan to use MySQL storage, you should also read the "MySQL storage" document.

A complete list of improvemenst can be found here: http://queuemetrics.loway.it/news.jsp
The latest version of QM can be downloaded from http://queuemetrics.loway.it/download.jsp

Yours,
l.

[VOIPSEC] Skype uncovered – Security study of Skype

Lots of skype papers these days... here's another one..

<http://www.ossir.org/windows/supports/2005/2005-11-07/EADS-CCR_Fabrice_Skype.pdf>

[VOPSECURITY] Interesting SIP DoS Attack

During the last IETF meeting an interesting draft was presented that I
thought can be a good discussion for the list and some of you may find
interesting.

In summary the attack involves forking SIP requests which cause the
generation of voluminous traffic that can cripple the affected SIP
proxies.

This is similar to an older attack which some of you may remember as
"finger-wars".

http://www.ietf.org/internet-drafts/draft-lawrence-maxforward-problems-00.txt

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