Digium Launches Echo Cancellation Cards

LINK: http://www.lightreading.com/document.asp?doc_id=75224&site=supercomm&WT.svl=wire1_2

Digium Launches Echo Cancellation Cards
JUNE 07, 2005

HUNTSVILLE, Ala. and CHICAGO -- Digium Inc., the creator of open source
telephony today announced the availability of two new echo cancellation cards,
the TE406P and the TE411P for Asterisk, the industry¿s first and most widely
used open source PBX. Asterisk offers a strategic, highly cost-effective
approach to voice and data transport over TDM, switched, and Ethernet
architectures.

The TE406P (for use with a 5.0-volt PCI slot) and the TE411P (for use with a
3.3-volt PCI slot) support both E1 and T1 environments and are selectable on a
per-card or per-port basis. The TE406P and the TE411P will support ultimate
density and performance with echo cancellation for four full T1 (96 channels)
or E1 (124 channels) and improves voice quality in situations where software
echo cancellation is not sufficient, is not done at the CO, or where CPU
utilization must be minimized. The cards are designed to perform in the most
difficult of environments while providing capacity/length trade-off by
supporting 16ms of echo cancellation over 128 channels, 32ms over 64 channels,
or 64ms over 32 channels.

"The combination of Digium hardware and Asterisk software enable numerous
combinations and configurations," said Mark Spencer, president of Digium and
the creator of Asterisk. "Our new cards will scale better while improving the
system's density and voice clarity."

New astGUIclient version released 1.1.1

Hello,

We've released another update to our Asterisk GUI Client suite: 1.1.1

http://astguiclient.sf.net/

Screen shots: http://astguiclient.sourceforge.net/screenshots.html

The client suite runs on both Windows and UNIX, includes the VICIDIAL
auto-dialer and 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 or IAX trunks.

In addition to many bug fixes and new features, we've added a web-based
version of the astGUIclient that only requires a web browser to use and
allows you to extend the functionality of your phone wherever you are.

Here are some of the features of the astGUIclient web client:
- Grabs live call info from a DB updated every second
- Displays live status of users phones and Zap/IAX/SIP/Local channels
- Allows calls to be placed from GUI and directed to phone
- Allows intrasystem calls at the click of a button
- Allows call recording at the click of a button
- Allows conference calling through GUI
- Administrative Hangup of any live Zap/IAX/SIP/Local channel
- Administrative Hijack of any live Zap/IAX/SIP/Local channel
- Administrative switch user function
- Call Parking sends calls to park ext and then redirects to phone ext
- CallerID popup with links to open custom web pages
- Voicemail display and button to go right to check voicemail
- Allows Blind transfers of calls to specific voicemail boxes
- Allows Blind transfers of calls to intrasystem extensions
- Allows Blind transfers of calls to external numbers
- Send to Voicemail directly from the inbound call popup window without
answering
- All client phone connections are shown not just the first
- Allows transfers to conferences
- Call parking with callerID

To upgrade from 1.1.0 you need to update all of the server apps and web
pages and run the update sql script.

Let me know what you think.

Thanks,

MATT---

Asterisk Virtual Configuration Testbed released

link: http://asterisk.ochsnet.com

Since I had a few requests for the web interface, I packaged it up
with a quick install guide and put it on http://asterisk.ochsnet.com.
It also includes the latest version of VConfig that now has support
for static database and realtime extensions all at once (you can use
one, both, or a combination all at the same time).

Just a warning, the docs are very slim. If you are good with apache
and have used mod perl, then you should be ok. I'll put up some
better documentation on the website when I can.

Chris

Asterisk LDAP Module now available for testing

Fellow developers,

My organization uses LDAP extensively to manage our customer information
and we were looking for a way to store phone system information with the
rest of our data. Unfortunately the solutions we found were not
adequate for our needs. To fill that void, I humbly submit
Asterisk::LDAP.

Asterisk::LDAP is a Perl module I have written and released under the
terms of the GPL which is capable of generating Asterisk version 1.0
compatible configuration files from an LDAP data source. Currently it
supports generation of extensions.conf, voicemail.conf and
musiconhold.conf. Future plans include sip.conf, iax.conf and
meetme.conf. As this is the first release it is sure to include lots
bugs so I disclaim any kind of responsiblity for anything including
remembering why I wrote any particular piece of code any particular way.
That being said it has been tested in my environment and I think it
works well. Example scripts are in the tarball, including one which can
be dropped in voicemail.conf's 'externpass' argument to update a user's
voice mailbox PIN when changed via the telephone interface.

There is much work left to do, not the least of which is documentation.
I have made an attempt to document the methods via perldoc ('perldoc
Asterisk::LDAP' after installation) and through the example code. I
welcome any suggestions or contributions.

I also want to say that I KNOW this is a big ugly hack. The Right Way
to do it will be Real Time Asterisk. Unfortunately I needed a solution
yesterday and this fit the bill.

If you think you'd like to take a look for yourself, hop on over to

http://projects.alkaloid.net.

Enjoy! Comments and criticisms to ben@alkaloid.net. Complaints
to /dev/null.

Thanks,
/BAK/

-- Ben Klang, KF4WBX Alkaloid Networks ben@alkaloid.net http://projects.alkaloid.net

E911 Services for VoIP Providers offered by DASH911.com (USA)

http://www.prleap.com/pr/8803/
E911 Services for VoIP Providers offered by DASH911.com

(PRLEAP.COM) 6 June 2005 Miami, FL - Dash911, an innovative technology
software engineering company, will begin offering E911 capability to
VoIP providers nationwide in the USA.

The new FCC ruling requires that all Internet broadband telephone
providers (VoIP providers) must offer E911 service to their
subscribers by October 2005. However, providing these services can be
expensive in terms of implementation effort, hardware, software,
dedicated lines and increased overhead expenses.

Dash911 offers VoIP providers a low-cost method of immediately
offering E911 services to broadband telephony subscribers through
pre-engineered server plug-ins, and private label subscriber-facing
web pages branded with the VoIP provider's name and logo.

The Dash911 system of VoIP E911 services offers nationwide USA
coverage and is fully-compliant with the FCC ruling and will provide a
more advanced solution that the commonly-referred to "ten digit
redirect".

The Dash911 system is engineered to be triple-redundant with
high-availability, carrier class, failover and fault-tolerant servers
in multiple geographic locations. For extreme emergency situations due
to any unlikely system problems there is a live person and
manually-handled call routing backup feature to ensure 100% E911 call
completion. To complete the offering, Dash911- enabled VoIP providers
will also be covered by a $1,000,000 insurance policy in relation to
E911 emergency calls.

In an effort to help startup and smaller VoIP providers be
competitive, the Dash911 VoIP E911 system is pre-engineered to work
with all Asterisk-based VoIP provider systems without any
modifications. A SOAP interface API is available for developers for
custom applications.

Prices for implementing Dash911's VoIP E911 service start at a
one-time $1,245 startup fee plus $0.68 to $1.08 per DID per month.
Complete details and signup information available at www.Dash911.com .

For more information, contact Michael Giagnocavo, CTO at info@ Dash911.com .

Details on the FCC ruling are at: www.fcc.gov

Digium and Cepstral Announce Text-To-Speech Partnership for Linux Telephony

link:http://press.arrivenet.com/tec/article.php/648853.html
Distribution Source : Market Wire
Date : Monday - June 06, 2005

PITTSBURGH, PA and HUNTSVILLE, AL -- (Market Wire - Jun 06, 2005) -- Digium
and Cepstral announced a partnership agreement today to provide Cepstral Swift
Text-To-Speech (TTS) languages and voices for Digium's Linux telephony
platforms and applications. Under the arrangement, Cepstral becomes a Digium
Premier Partner for providing commercial Text-To-Speech.

The offering will include Cepstral's telephony voices, David, Diane and William
for U.S. English, as well as options for U.K. English, German, Canadian French,
American Spanish, and Italian, for use in multi-channel applications on the
Linux operating system. Each will be available from Digium, and incorporated
into key products and services.

"Cepstral voices allow our customers to introduce high quality Text-To-Speech
into telephone and IP telephony applications that use dynamic content. These
voices sound great and provide the flexibility to deliver information over the
phone, or on the network," said Mark Spencer, CEO of Digium and primary author
of the Asterisk PBX.

"Digium is a proven pioneer who's development of the Asterisk PBX has created a
powerful and practical platform for computer-telephony integration," said Kevin
Lenzo, CEO and co-founder of Cepstral, LLC. "We are very happy to partner with
the company that is spearheading the Linux telephony movement. Our new
server-based voices fit the offering perfectly and bring extraordinary vocal
quality at an extremely competitive price."

Digium's products and services featuring the Cepstral Swift TTS voices will be
available in June of 2005.

About Cepstral, LLC

Cepstral is a speech technology company based in Pittsburgh, PA, USA, which
provides speech technologies and services for the spoken delivery of
information. Cepstral builds high quality, natural sounding voices for server,
desktop, and hand-held applications. Cepstral also provides professional
services to customize and tune voices. Cepstral: We Build Voices(TM).

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 implementations of PBX, IVR, media gateway, and
communications servers through open source, standards-based software and
innovative hardware solutions. Digium also offers technical support and
development services for Asterisk, and commercial versions such as Asterisk
Business Edition, a professional-grade distribution of the acclaimed
open-source software.

Contact:
Craig Campbell
Cepstral, LLC
412/432-0400 Ext. 604
press@cepstral.com

http://www.cepstral.com/

« Previous Page