We are a UNIX networking company established in 1994 to service the Quinte area. We specialize in UNIX BSD networking and Internet connectivity.
VOIP
Skype Introduces 10-Way Video Calling
Skype 5.0 beta two is already available for download; it includes 10-way video calls, automatic call recovery and a cleaner user interface. The update is also said to improve call quality and includes a number of bug fixes to make the overall experience much smoother.
Crestron offers new MTX-3 wireless touchpanel remote
The MTX-3 offers seamless interaction with AV and environmental systems, providing true feedback of all settings, and displaying metadata information for all digital media. Crestron’s infiNET EX wireless technology provides reliable two-way communications throughout a residence or commercial structure utilising a 2.4 GHz mesh network.
Cisco making a play for Skype?
Cisco is reportedly looking to buy Skype before the Internet phone provider goes public. The blog TechCrunch posted over the weekend that Cisco made an offer for Skype before it completed its IPO process. The site attributed the unconfirmed information to "reliable sources."
Taiwan government authorities astounded at Intel WiMAX move
FCC's Closed-Door Net Neutrality Meetings Break Down
Meet The 2600hz Project, The New Sound of Open Source Telephony
Gigaom - Some of the core developers behind FreePBX — a well-known, open-source phone system — have teamed up and started The 2600hz Project, a commercial entity promoting a collection of open-source telephony applications and libraries. Today, they are releasing blue.box, a reworked version of open source FreePBX. The new venture is co-founded by Darren Schreiber and is also a subsidiary of newly formed VoIP Inc. The 2600hz Project received $250,000 in funding from an unnamed investor.
Asterisk PBX 1.6.2.10 Now Available
he Asterisk Development Team has announced the release of Asterisk PBX 1.6.2.10. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.10 resolves several issues reported by the community and would have not been possible without your participation.
The following are a few of the issues resolved by community developers:
* Allow users to specify a port for DUNDI peers.
(Closes issue #17056. Reported, patched by klaus3000)
* Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is
set.
(Closes issue #16815. Reported, patched by rain)
* If there is realtime configuration, it does not get re-read on reload unless
the config file also changes.
(Closes issue #16982. Reported, patched by dmitri)
* Send AgentComplete manager event for attended transfers.
(Closes issue #16819. Reported, patched by elbriga)
* Correct manager variable 'EventList' case.
(Closes issue #17520. Reported, patched by kobaz)
In addition, changes to res_timing_pthread that should make it more stable have
also been implemented.
For a full list of changes in the current release, please see the
ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.10
Thank you for your continued support of Asterisk!
Asterisk PBX 1.4.34 Now Available
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.34 resolves several issues reported by the community and would have not been possible without your participation. Thank you!
The following are a few of the issues resolved by community developers:
* Allow users to specify a port for DUNDi peers.
(Closes issue #17056. Reported, patched by klaus3000)
* Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is
set.
(Closes issue #16815. Reported, patched by rain)
* First caller into a dynamic conference new enters the pin once.
(Closes issue #15878. Reported, patched by pabelanger)
* Send AgentComplete manager events in the event of blind and attended
transfers.
(Closes issue #16819. Reported, patched by elbriga)
For a full list of changes in the current release, please see the
ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.34
Thank you for your continued support of Asterisk!
Cisco releases new tablet called Cisco Home Energy Controller for smart home market
Tablets or just gadgets in the broad sense that do our work is finally coming close to being a reality especially with the Cisco tablet that has been released recently. Seated at just one single area of the house, the tablet provides the user with complete control of all the other utility items in the house.
Named the Cisco Home Energy Controller, the tablet is an even more specialized product from Cisco that comes hot on the heels of the other tablet offering from the same company in the form of the Cius that has been built purely with a business scenario in mind.
U.S. has 21 million VoIP subscribers according to FCC report
The U.S. Federal Communications Commission reports that there are 21 million VoIP subscriptions stateside, and that the vast majority of them are residential customers. But it didn't count Skype subscribers.
These figures were announced by the FCC to accompany the release of its highly detailed 31-page report, "Local Telephone Competition". The report, however, isn't exactly what you might call "fresh" — although it was released just last Friday, its data set ends on December 31, 2008.
Video Conferencing's Coming Boom: Polycom CEO Predicts Rapid Growth
Asterisk libpri 1.4.11.3 Now Available
http://downloads.asterisk.org/pub/telephony/libpri/
This release fixes a regression in the calling number assignment logic:
* Calling Number assignment logic change in libpri 1.4.11. Restored the old behaviour if there is more than one calling number in the incoming SETUP message. A network provided number is reported as ANI.
(Closes issue #17495. Reported and tested by ibercom. Patched by rmudgett)
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.4.11.3
Thank you for your continued support of Asterisk!
Asterisk PBX 1.4.33.1 Released
The Asterisk Development Team has announced the release of Asterisk 1.4.33.1. This release is available for immediate download at:
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.33.1 resolves a regression involving the use of FXO signaling in chan_dahdi where a channel could continue ringing after it has been answered.
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.33.1
Thank you for your continued support of Asterisk!
EB Releases Hardened IP VoIP Phone for Military & Defence Use
Elektrobit Delivers Its Hardened VoIP Solution to Finnish Army
EB Tough VoIP will operate under demanding environmental conditions for the Army branch of the Finnish Defence Forces. It will work with the Finnish Army on various testing phases throughout this year “in order to ensure proper usage and meet system requirements.”
Asterisk PBX 1.6.2.8 Now Available
The Asterisk Development Team has announced the release of Asterisk PBX 1.6.2.8. This release is available for immediate download at:
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.8 resolves several issues reported by the community, and would have not been possible without your participation.
The following are a few of the issues resolved by community developers:
* Enable auto complete for CLI command 'logger set level'.
(Closes issue #17152. Reported, patched by pabelanger)
* Make the mixmonitor thread process audio frames faster.
(Closes issue #17078. Reported, tested by geoff2010. Patched by dhubbard)
* Add missing 'useragent' field to sip-friends.sql file.
(Closes issue #17171. Reported, patched by thehar)
* Add example dialplan for dialing ISN numbers (http://www.freenum.org)
(Closes issue #17058. Reported, patched by pprindeville)
* Fix issue with double "sip:" in header field.
(Closes issue #15847. Reported, patched by ebroad)
* Add ability to generate ASCII documentation from the TeX files by running
'make asterisk.txt'.
(Closes issue #17220. Reported by lmadsen. Tested, patched by pabelanger)
* When StopMonitor() is called, ensure that it will not be restarted by a
channel event.
(Closes issue #16590. Reported, patched by kkm)
* Small error in the T.140 RTP port verbose log.
(Closes issue #16998. Reported, patched by frawd. Tested by russell)
For a full list of changes in the current release, please see the
ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.8
Thank you for your continued support of Asterisk!
--
Snom releases the 821 IP Phone that supports gigabit ethernet.
What Exactly Is '4G'?
Skype Really Wants Your Business
Cisco takes wraps off home TelePresence solution
It's not just Google who has taken the leap into consumer appliances. Just the day after the search company launched the Google TV, Cisco has announced trials of a consumer version of its TelePresence videoconferencing system according to a Reuters report. The company has also announced that TV channel ESPN will be using Telepresence to transmit World Cup football from South Africa.
In January, the company announced that it would be running trials of a home Telepresence system with Verizon in the US and, later in the year, France Telecom.
