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Digium Inc., the Asterisk company, today announced Asterisk 1.4. According to Digium, the new version isn't available for download until October however. Although I know Mark Spencer, I have no inside information, however, my guess is that Asterisk will launch the software at TMC's
Internet Telephony Conference & Expo in San Diego. Seems like the perfect venue to launch this major software release.
Asterisk 1.4 is the first major release of Asterisk since the release of Asterisk 1.2 in November 2005. It includes over 20 new functionality additions including IPFAX compatibility, unified messaging capabilities and
Jabber/Jingle/GoogleTalk protocol compatibilities. Asterisk 1.4 features overall quality and performance improvements, as well as increased scalability and interoperability.
"This is by far the best version of Asterisk to date,” said Mark Spencer, president of Digium and creator of Asterisk. "With the support of the Asterisk community, we have been able to develop an advanced platform that will make it even easier for users to migrate to VoIP, especially those in the enterprise community.”
Specific enhancements featured in Asterisk 1.4 include:
• Generic Jitter Buffer- improves the quality of a call during network congestion.
• Asterisk Extension Language Version 2- simplifies programming and dial plan configuration.
• T.38- allows IP FAXes to pass through the server.
• Jabber/Jingle/GoogleTalk- supports compatibility with all of these networks.
• Increased language capabilities- offers new language capabilities in English, Spanish and French as well as new sounds and improved sentence structure support.
• Unified Messaging- integrates voicemail, email, and fax into a central mailbox where users can send, retrieve and manage all of their messages using any communication device.
• Whisper Paging- allows for selective, pre-programmed call interruption with controlled volume levels and muting capabilities.
Additionally, Asterisk 1.4 now includes variable length DTMF support (touch-tone signaling for IVR applications), the option for programming shared line appearance, centralized RADIUS storage for call detail records, a built-in web manager interface and a simplified, single user configuration for SOHO/SMB users. Asterisk 1.4 also offers increased memory usage and performance improvements such as improved interoperability of SIP call transfers, IAX2 scalability improvements, enhanced IAX2 media stream capabilities (enabling direct audio communication between IAX devices while eliminating server involvement and maintaining billing and control functionalities), Cisco® SCCP support, SNMP monitoring, and RTP native bridging capabilities.
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Written by VoIP & Gadgets Blog on September 12th, 2006 with no comments.
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Andy had the
goods yesterday on Sixapart's Live Journal and
Gizmo Project in a deal that would enable one of the largest blogging community to be able to IM & VoIP each other. Well, today, it's official.
The co-branded “Gizmo Project for LJ Talk” software will enable LiveJournal friends to call each other for free, send instant messages and see each other's online presence (available, away or offline). Some nice features include free “Call me” and “IM” buttons on journal pages to initiate calls and instant messages from the web, as well as point and click “voice posts", which allows Gizmo Project for LJ Talk users to post audio recordings (podcasts) on their blog. The partnership also stated that Gizmo Project for LJ Talk users will be able to make low-cost PSTN calls in addition to buddy-to-buddy PC calling. Gizmo leverages the SIP standard for outbound dialing.
Six Apart today simultaneously announced it has created an open source Jabber server, a modular, scalable, server that enables others to integrate their services with Jabber and LJ Talk, LiveJournal's new instant messaging and voice service. The new server provides smoothly integrated IM and voice capabilities, and is now available for any company or service provider to use for free.
Google Talk, which is Jabber-based is able to connect to Six Apart's Jabber server. Similarly, Gaim, Adium, iChat, Trillian Pro, and Pandion S, all Jabber-based should be able to connect as well.
Are Skype's proprietary days numbered? They don't use SIP, they don't use Jabber, their video is proprietary, and they don't allow other client's to connect.
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Written by VoIP & Gadgets Blog on July 27th, 2006 with no comments.
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Remember my
in-depth overview of EQO Communications and their mobile-to-VoIP (Skype), mobile-to-IM (GoogleTalk, AOL, Skype, etc.) and mobile-to-social communities software?
Well, I had word last week that tomorrow
EQO Communications will announce support for RIM Blackberry and Windows Mobile powered devices. With this latest release, Blackberry and Windows Mobile users will be able to access EQO Mobile, a platform that enables VoIP calling and instant messenger services (including AOL AIM, ICQ, GoogleTalk, Jabber, MSN, Yahoo! and Skype) on mobile devices.
According to EQO, the addition of handsets such as the Blackberry Pearl and Treo 700w brings the number of EQO-supported handsets to more than 400 mobile devices, including Nokia, Motorola, Sony Ericsson, Palm, LG, Panasonic, Phillips, Siemens, Sanyo, Sharp and Samsung.
Although I already covered EQO extensively in a
prior post, essentially EQO gives you click-to-call mobile VoIP and click-to-chat mobile IM using a patent pending core IP that uses proprietary call signaling that is P2P-based and cross device/network domain. It leverages mobile data (SMS) to control the signalling and the mobile voice network for the media stream to the mobile phone, so for instance you don't need the Skype, AOL, GoogleTalk, etc. clients running on your cell phone.
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Written by VoIP & Gadgets Blog on January 1st, 1970 with no comments.
Read more articles on Skype and VoIP and Mobile Phones and yahoo and AOL and mobile phone and msn and im and icq and jabber.
I recently discovered
Hotty.com, a website that is offering a new SIP VoIP Softphone,
Megapin Inc. has developed a few SIP and Jabber products that they offer to service providers and wholesalers to OEM, with full "skinning" capabilties. One of their prodis
M2-UA, a SIP softphone application and the other is
M2-IM, which has both SIP and Jabber IM/presence support. The M2-IM product will support
Yahoo! Messenger, MSN Messenger, AIM, ICQ, and
Google Talk. Their softphone products feature standard voice codecs including G.711, GSM, and Speex.
I was actually checking out the Hotty.com website last week, but now their homepage shows a 55 day countdown till the product launches. Where'd the content I saw last week go? Fortunately, I keep a very lengthy browser "cache", so I was able to copy the screenshots from my cache folder.
I still haven't figured out why Megapin, Inc. is using the domain name
hotty.com for a softphone application. Sounds like a porno site if you ask me. And if you look at the black background with the coundown clock it certainly has the feeling of a porn site to me. Either that or a hacker site, which often use a black background. Black is cool in hacker world. Maybe their softphone application will be targetted at online dating sites or perhaps sites with female "
hotties" taking inbound VoIP calls for some Phone Sex over IP (after you give your credit card number of course)
Well, only 55 more days till we find out what this is all about...
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Written by VoIP & Gadgets Blog on January 1st, 1970 with no comments.
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