[asterisk-users] Announcement: Asterisk Service Provider Edition v1.0 Beta

[Olle E Johansson]

The Asterisk Developer Team is proud to announce the Asterisk SPE  
v1.0 Beta Release
for immediate download on tftp.digium.com.
The SPE has been developed as a joint project between Digium, the  
Asterisk Company,
Voop, the European Asterisk Dialtone provider and the Asterisk  
community.

The Asterisk Service Provider Edition is focused on the needs for the  
new breed
of Telecom companies - the Voice over IP Service Providers.  It will  
be available
both as a free download in Open Source and as a commercial product
called Asterisk Commercial Service Provider Edition, ACSPE.

- "We felt the need to focus on being an enabler for this new kind of  
telco,
    making sure that Asterisk fits into their network as well as  
business models
    in a professional way" says Matt Penser, Asterisk innovator. "The  
previous versions
    was more targeted to the needs of the business user, a market where
    Asterisk already is stronger than any other offering on the market".

The Asterisk SPE has a number of new features, that makes it the most  
powerful
platform for these companies. No other Open Source package can deliver a
matching feature set:

- All the features from Asterisk 1.4 and the business edition
- Asterisk VoipRoute(R) technology for SmartRTP(R) bridging
- Asterisk RateRoute(TM) technology for route selection
- Asterisk SpitWall(R) core for SPIT filtering

These new solutions will enhance Asterisk and will help the VSP's to
leap lightyears ahead of their competion.

* Asterisk VoipRoute(R) SmartRTP(R) Bridging
----------------------------------------------------------------

The VoipRoute SmartRTP bridging technology enhances the Asterisk RTP
bridge with a new scheme. In addition to the current RTP bridges -  
the native bridge,
the remote bridge and the hybrid RTP-direct bridge, SmartRTP uses a  
combination
of the BGP IP routing protocols and the TRIP VoIP routing system to  
find the
best and fastest way to route calls between IP nodes on the Internet  
or local network.

- "The SmartRTP bridge system, based on our patented VoipRoute core,
    makes sure that call latency is minimal. We also enhanced it with a
    MediaRescue solution that will capture lost media frames and re- 
insert
    them in the audio or video stream before it reaches the  
destination." says
    Josua Polk, the Asterisk RTP developer.
    "This system implements an Asterisk VoipRoute layer on top of the  
Internet
    and uses Dundi(TM) to automatically discover new SmartRTP relays  
and their
    properties. It practically erases packet loss, jitter and latency  
from the list of
    issues for the provider's support department. We call it SPEake- 
friendly calls!"


* Asterisk RateRoute(TM) Least Cost Routing
-------------------------------------------------------------

The RateRoute(TM) solution is only available in the ACSPE due to  
licenses from
other vendors, soon to be disclosed. The RateRoute system analyze the  
call
from fifteen distinct properties and use an external hardware  
accelerator to
find the best route to forward the call, be it PSTN or VoIP channels.  
By using
the hardware accelerator RR520P PCI express card, LCR decisions is now
down to microseconds without accessing external databases.

- "We've implemented this in our commercial VoIP network during  
development,
    and cut our costs by at least 75% and enhanced call quality.  
Billing and CDR
    mediation is much easier, since the RateRoute system always  
picked one
    outbound service provider that always matched the fifteen  
criteria for
    carrier selection" says Anders Runnstam at PulseVoip in Bergen,  
Norway.


* Asterisk SPITwall(R) - filtering away tomorrows VoIP spam today!
------------------------------------------------------------------------ 
------------------

The SPITwall(R) technology is developed by Olle E. Johansson, a
member of the Asterisk developer team and Senior Technical Advisor
for Voop in Bergen, Norway - the Asterisk Dialtone provider.

- "I got more and more annoying calls during development, which  
disturbed
   me a lot and caused me to loose concentration. On the other hand,
   it inspired me to develop SPITwall to be able to filter them out.
   I have measured up to 95% success rate on call filtering,
   which is far beyond any similar products on the market. By not  
bothering
   with answering the final 5%, I could concentrate on development again
   and succesfully finish my development projects." says Olle.

The SPITwall is built on a shared database and use bayesian techniques
to analyze the content of the call. It requires Asterisk ChanSpy to  
be able
to listen in and warn the callee about ongoing unsolicited calls. The  
callee
can also press certain DTMF sequences during the call to mark the call
as SPIT. The voice pattern, SPITwall checksums and call properties will
then immediately be stored in the Digium SPITcore repository to be
available for all other users.

- "Using the community to build a SPIT-fighting database is natural for
   an Open Source project. The community is the power of Asterisk and
   by sharing a resource like this, we can make sure that everyone
   contributes. The SPITshare(r) analyzer makes sure that companies
   that does not contribute will get older data and more SPIT calls"  
says
   Jill Timmer, VP or marketing.

SPITwall 1.0 is available with English, Norwegian and Swedish language
support. Some support for Canadian and southern US dialects is
implemented, and will be finished by release time.

                                            --- o ---

In addition to these revolutionary features, Asterisk Service  
Provider Edition
will contain full T.39 faxing (leapfrogging systems that only support  
up to
version 38), the Codename Pomengranade SIP Stack, the Project Okapi
IAX3 trunking technology featuring IAX3 over SS7 transport, the VoipVote
digital phone voting system with app_preselect cheating technology
and a fully working version of the VirtuAST virtual PBX hosting Asterisk
virtualisation core.

Asterisk SPE v1.0 beta is available for immediate download. At this
time we're looking for feedback from service providers.

Release date for the 1.0 version is to be released, pending beta tests.
ACSP Edition will be available with  Telco level 24/8 support
May 15th, 2007. The RR520 RateRoute hardware accelerator is in
distribution through authorized resellers starting April 10th.

Asterisk, Digium, SPITwall, SpitShare, VoipRoute, SmartRTP, RateRoute
and Dundi are trademarks that may be registered by Digium, Inc.

For immediate release, April 1st 2007
On behalf of the Asterisk development team and project 0401

/Olle