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September 30, 2010

Patent Office Says Another 'Worst Patent' Should Be Rejected As Obvious

The latest news in the ongoing effort by the EFF to invalidate ten awful patents looks good, as the Patent Office has given an initial rejection of C2's VoIP patent, claiming that it qualifies as "obvious." The incredibly broad patent (6,243,373) basically covers all VoIP implementations. Of course, this is just the "first office action," which rarely means very much, since the company still has the ability to come back and beg and plead for the USPTO to keep the patent alive (which happens often enough).
Still, it does make you wonder, since it certainly does seem like it was an abundantly obvious patent (yes, even back when it was filed -- someone should talk to Jeff Pulver for some prior art), why it's taken this long for the USPTO to begin to correct its error.
 
While we continue to applaud the EFF for working to get these patents busted, as we mentioned recently, the real travesty is that it's been six years since the EFF began busting patents, and while there's progress on nearly all of the patents, it's an incredible slog -- and these are for the worst of the worst patents. Invalidating bad patents is a ridiculously difficult process. That's really bad, considering all the harm they can do in the meantime.
 

Posted by Dal at 03:26 PM | | TrackBacks (0) Posted to VoIP News

September 23, 2010

Asterisk PBX 1.8.0 Release Candidate 2 Now Available

The Asterisk Development Team has announced the second release candidate of Asterisk PBX 1.8.0. This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

Asterisk 1.8.0-rc1 was not released due to an issue found prior to release.

 * Make AMI honor enabled=no
   (Closes issue #18040. Reported by: twilson
    Review: https://reviewboard.asterisk.org/r/938/)

All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker,

https://issues.asterisk.org/. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list.

Asterisk 1.8 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. For more information about support time lines for Asterisk releases, see the Asterisk versions page.

http://www.asterisk.org/asterisk-versions

With the availability of the Asterisk 1.8.0 release candidates, the binary add-on modules for Asterisk produced by Digium have been updated with new versions that are compatible with Asterisk 1.8. The availability of these
modules will assist with the testing of Asterisk 1.8.0 in a wider variety of situations.

This release candidate contains fixes since the last beta release as reported by the community. A sampling of the changes in this release candidate include:

 * Add slin16 support for format_wav (new wav16 file extension)
   (Closes issue #15029. Reported, patched by andrew. Tested by Qwell)

 * Fixes a bug in manager.c where the default configuration values weren't reset
   when the manager configuration was reloaded.
   (Closes issue #17917. Reported by lmadsen. Patched by bbryant)

 * Various fixes for the calendar modules.
   (Patched by Jan Kalab.
    Reviewboard: https://reviewboard.asterisk.org/r/880/
    Closes issue #17877. Review: https://reviewboard.asterisk.org/r/916/
    Closes issue #17776. Review: https://reviewboard.asterisk.org/r/921/)

 * Add CHANNEL(checkhangup) to check whether a channel is in the process of
   being hung up.
   (Closes issue #17652. Reported, patched by kobaz)

 * Fix a bug with MeetMe where after announcing the amount of time left in a
   conference, if music on hold was playing, it doesn't restart.
   (Closes issue #17408, Reported, patched by sysreq)

 * Fix interoperability problems with session timer behavior in Asterisk.
   (Closes issue #17005. Reported by alexcarey. Patched by dvossel)

 * Rate limit calls to fsync() to 1 per second after astdb updates. Astdb was
   determined to be one of the most significant bottlenecks in SIP registration
   processing. This patch improved the speed of an astdb load test by 50000%
   (yes, Fifty-Thousand Percent). On this particular load test setup, this
   doubled the number of SIP registrations the server could handle.
   (Review: https://reviewboard.asterisk.org/r/825/)

 * Don't clear the username from a realtime database when a registration
   expires. Non-realtime chan_sip does not clear the username from memory when a
   registration expiries so realtime probably shouldn't either.
   (Closes issue #17551. Reported, patched by: ricardolandim. Patched by
    mnicholson)

 * Don't hang up a call on an SRTP unprotect failure. Also make it more obvious
   when there is an issue en/decrypting.
   (Closes issue #17563. Reported by Alexcr. Patched by sfritsch. Tested by
    twilson)

 * Many more issues. This is a significant upgrade over Asterisk 1.8.0 beta 5!

A short list of available features includes:

 * Secure RTP
 * IPv6 Support in the SIP channel driver
 * Connected Party Identification Support
 * Calendaring Integration
 * A new call logging system, Channel Event Logging (CEL)
 * Distributed Device State using Jabber/XMPP PubSub
 * Call Completion Supplementary Services support
 * Advice of Charge support
 * Much, much more!

A full list of new features can be found in the CHANGES file.

http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout

For a full list of changes in the current release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc2

Thank you for your continued support of Asterisk!

Posted by Dal at 04:12 PM | | TrackBacks (0) Posted to Asterisk Releases

AireSpring announces VoMPLS (VoIP over MPLS)

SIP Trunking provider AireSpring (Booth #3016) announced at the Channel Partners Conference & Expo the launch of its new Managed Voice (VoIP) over MPLS product (VoMPLS). Available nationwide across three tier-one networks, AireSpring MPLS-VPN targets multi-location businesses requiring secure, flexible, and intelligent voice and data connections. This solution, the company says, offers maximum voice and data bandwidth and security at the most economical rates. 

By combining managed IP voice services with an MPLS network, AireSpring says it delivers the next generation of communications technology today. In addition, AireSpring can hand off voice capacity to the customer via its preferred option including digital T1/PRI, analog POTS lines or SIP trunks.  Airespring VoMPLS can be customized with a full suite of flexible calling features and configuration options. From the smallest SMB’s to the largest enterprises, AireSpring says its Voice Over MPLS is an affordable and attractive solution.

“Our customers and agents are increasingly looking to update their voice and data networks with the latest secure, managed voice and data technology," said AireSpring COO Daniel Lonstein.  “One of the great advantages to AireSpring voice services is the adaptability of the product to the customer’s current and future phone system. Because our voice network is all IP from the customer premise, we can easily convert a customer’s connection from T1/PRI or analog to SIP trunking without requiring a new installation.

We like to call our voice services “future-proof." And with the unparalleled footprint of three separate underlying Tier 1 MPLS networks, we can reach nearly every potential MPLS location in America. This is huge news for us and for our customers and agents."

Source

Posted by Dal at 04:07 PM | | TrackBacks (0) Posted to VoIP News

Cisco revamps Flip UltraHD and MinoHD camcorder lines

While the new line of Flip camcorders don’t look radically different in an aesthetic sense, they do increase the product line’s technical abilities and consumer appeal while revamping the high-definition functionality of the UltraHD and MinoHD platforms.

More pointedly, the two-hour UltraHD comes with the new FlipPort accessory connector and can record video content in 720p at a frame rate of 60 frames per second (fps), which is a significant improvement over the 30fps rate held by the one-hour version.

Both of the new MinoHD devices record at 720p and 60fps.

In a move to make Flip video capture even smoother than ever before, the newly tweaked devices also come equipped with enhanced digital image stabilisation technology to cut down on unwanted camera shake.

Other features evident on all the new cameras include the usual flip-out USB connector, video capture via MP4, an HDMI-out port for playback on a high-definition television, and the proprietary FlipShare software platform.

The 8GB UltraHD Flip comes with a two-hour rechargeable battery and can also run on AAA batteries, while the 4GB UltraHD offers one hour of operation via AAA batteries. 

The two new UltraHD additions are priced at $200 USD for the 8GB model and $150 USD for the smaller 4GB version. The 8GB MinoHD costs $230 USD, while the 4GB model costs $180 USD.

Source

Posted by Dal at 04:04 PM | | TrackBacks (0) Posted to Cisco

September 15, 2010

Asterisk PBX 1.6.2.12 Now Available

The Asterisk Development Team has announced the release of Asterisk PBX 1.6.2.12.  This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.6.2.12 resolves several issues reported by the community and would have not been possible without your participation.

The following is a sample of the issues resolved in this release:

    * Fix issue where DNID does not get cleared on a new call when using
      immediate=yes with ISDN signaling.
      (Closes issue #17568. Reported by wuwu. Patched by rmudgett)
    * Several updates to res_config_ldap.
      (Closes issue #13573. Reported by navkumar. Patched by navkumar, bencer.
      Tested by suretec)
    * Prevent loss of Caller ID information set on local channel after masquerade.
      (Closes issue #17138. Reported by kobaz, patched by jpeeler)
    * Fix SIP peers memory leak.
      (Closes issue #17774. Reported, patched by kkm)
    * Add Danish support to say.conf.sample
      (Closes issue #17836. Reported, patched by RoadKill)
    * Ensure SSRC is changed when media source is changed to resolve audio delay.
      (Closes issue #17404. Reported, tested by sdolloff. Patched by jpeeler)
    * Only do magic pickup when notifycid is enabled.
      A new way of doing BLF pickup was introduced into 1.6.2. This feature adds a
      call-id value into the XML of a SIP_NOTIFY message sent to alert a subscriber
      that a device is ringing. This option should only be enabled when the new
      'notifycid' option is set, but this was not the case. Instead the call-id
      value was included for every RINGING Notify message, which caused a
      regression for people who used other methods for call pickup.
      (Closes issue #17633. Reported, patched by urosh. Patched by dvossel.
      Tested by: dvossel, urosh, okrief, alecdavis)

For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.12

Thank you for your continued support of Asterisk!

Posted by Dal at 11:07 AM | | TrackBacks (0) Posted to Asterisk Releases

Asterisk PBX 1.4.36 Now Available

The Asterisk Development Team has announced the release of Asterisk PBX 1.4.36. This release is available for immediate download at:
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.4.36 resolves several issues reported by the community and would have not been possible without your participation.

The following is a sample of the issues resolved in this release candidate:

    * Fix issue where DNID does not get cleared on a new call when using
      immediate=yes with ISDN signaling.
      (Closes issue #17568. Reported by wuwu. Patched by rmudgett)
    * Fix issue where SIP promiscuous redirect could fail to dial the
      redirect (app_queue).
    * Fixes issue with translator frame not getting freed. This issue prevented
      G.729 licenses from being freed up.
      (Closes issue #17630. Reported by manvirr. Patched by dvossel)
    * Ensure SSRC is changed when media source is changed to resolve audio delay.
      (Closes issue #17404. Reported, tested by sdolloff. Patched by jpeeler)
    * Q931 - Sending PROGRESS after sending ALERTING is a protocol error.
      (Closes issue #17874. Reported, patched by nic_bellamy)

For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.36

Thank you for your continued support of Asterisk!

Posted by Dal at 11:04 AM | | TrackBacks (0) Posted to Asterisk Releases

September 13, 2010

Suitors May Be Calling on Polycom

A technology sector that has attracted scant investor attention is videoconferencing, even though it's on its way to becoming the chosen tool for companies seeking efficiency and productivity with greatly reduced travel costs. The major Wall Street houses have all but ignored the group, so not much is known about stocks that excel here.

Cisco Systems, which recently acquired videoconferencing firm Tandberg, is the top provider of videoconferencing gear. But another key, if smaller player, in this rising market that has caught the eye of some investors is Polycom. It develops and makes video-, voice-, data- and Web-conferencing equipment systems that provide corporate users with a complete communications and videoconferencing network infrastructure.

But what adds spice to the stock is its potential as an attractive takeover target. Rumors have swirled that Polycom has been approached last year by two different groups with feelers proposing a merger or a buyout. Such speculation is fueled in part by Polycom's alliances to develop and market voice over Internet protocol (VOIP) communications products with several major tech companies, including Hewlett-Packard, Microsoft and Avaya, a unit of Nortel Networks.

In Microsoft's case, Polycom has a deeper relationship that involves a development and marketing pact to integrate its desktop, conference-room video system and network hardware and software systems.

Source

Posted by Dal at 02:57 PM | | TrackBacks (0) Posted to

September 03, 2010

Skype Introduces 10-Way Video Calling

Skype — apparently pleased with its five-way beta group video-calling functionality — has just released a new version of Skype 5.0 for Windows that doubles group support. It now allows for up to 10 video callers.

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.

Of course, the standout feature is 10-way video calling, something that certainly one-ups their own previous offering and makes it suitable for even larger virtual team meetings and mini family reunions. Of course, it also makes Gmail’s (Gmail) video-calling functionality look like the ugly step sister — a proactive move on the part of Skype (Skype) to combat recent buzz surrounding Gmail Voice Calling.

Still, Skype does caution that 5.0 is beta, and hence, very buggy. It’s also limited to Windows (Windows) users, and 10-way video calls require all group chatters to be using the same second beta version of the app. Have you tested out five-way video calls? Are you ready to upgrade to the 10-person variety?

Source

Posted by Dal at 01:44 PM | | TrackBacks (0) Posted to Skype

September 02, 2010

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.

A complete infiNET EX network uses the lighting dimmers and other devices throughout the structure as wireless relay stations, each receiving and passing on wireless commands to the central gateway.

Every device that is added to the network effectively increases the range, strength, and reliability of the entire network by providing multiple redundant signal paths, ensuring that every button press is executed instantly and consistently.

The MTX-3 can also communicate directly with the gateway if no other infiNET EX devices are installed. Up to six MTX-3s can be assigned to a single gateway.

Source

Posted by Dal at 03:38 PM | | TrackBacks (0) Posted to Wireless Hardware

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."

It would be a multibillion purchase as Skype is looking to raise $5 billion in its initial offer, according to TechCrunch.Cisco declined to comment. Skype was not immediately available for comment.

The acquisition would be key to Cisco's thrust into the unified communications and collaboration, and consumer markets. It would bring to the company what is now a free and consumer friendly voice and video capability to augment the IP telephony and unified communications systems it now provides to corporate enterprises.

Integrated with a device such as the Flip pocket videocamera -- which would need an Internet access connection like the expected Wi-Fi capability -- Cisco could offer a handheld voice/data/video device for the consumer and perhaps enterprise market.

Source

Posted by Dal at 03:36 PM | | TrackBacks (0) Posted to Skype

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