Gizmo Project Alive? Not any more, Like the Related Gizmocall

When it was a thing in 2007, Gizmo Project was a paid VoIP service with a dedicated client. Later the domain and the name took Gizmo5, gozmo5.com giz5.com etc. Just like the GizmoCall
Gizmo Project is a closed-source software phone developed by SIPphone. It became a free product at sometime. It was a VoIP phone client, made specifically for the SIPPhone network. It complied with Open Standards then and directly peered with several other VOIP providers. GizmoProject, Gizmo5 could also directly dial sip URIs. For example, if you enter: sip:613@fwd.pulver.com you can reach the echo test at free world dialup, the goto VoIP test destination, them
It ran natively on Mac, Windows and Linux. For Linux, Debian packages were provided at the website.
It features very high-quality codecs.

• NAT support: Worked behind routers and NAT
• Supported STUN where applicable
• Supported for Symmetric RTP
• Supported for SIP Outbound proxy
• Supported codecs were: GSM, iSAC, iLBC, g7.11U/A, EG7.11, iPCM
• Complied with the SIP standard
• Instant Messages using XMPP (“Jabber”) is supported in the latest version, including s2s.

Gizmo is a Free Phone for Your Computer

That makes calling as easy as instant messaging

Key Features

• Unlimited FREE calling
• FREE calling to landline and mobile phones using the All Calls FREE plan.
• Chat with Windows Live Messenger, AIM, Yahoo! Messenger, GoogleTalk, and Gizmo users.
• FREE voicemail
• More features...
•  

Three Steps to Free Calling to 60 Countries

Download Gizmo. Sign-up;
add all your phone
numbers to your profile

Tell your friends and
family to do the same

Call those contacts on
Gizmo, their mobile phone
or landline for free*

*More Details & List of Countries | Download Now!

Get your own local number

Get a local number in over 170 countries and let people call you at local rates. Your friends or business associates call a unique, local phone number that connects them directly to you, wherever you are. Learn More

Calling landlines and mobile phones

With Call Out credit call any Mobile or Landline phone in the world at ultra low rates. Buy credit as you need it - no contracts or commitments. Learn More

Call Out Rates
Not Free, But Pretty Close

Select a country:

Mobile Phones

1.9

cents per/min

Landlines

1.9

cents per/min

 

Asterisk 1.8.0 Released

#Asterisk18 http://snapvoip.blogspot.com/
The Asterisk Development Team has released the latest and the greatest (so far) version of Asterisk
1.8.0. This release could be downloaded from Asterisk.org.


Asterisk 1.8 is the much anticipated version in the major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. The Asterisk versions page has more information about support time lines for Asterisk releases,
The release of Asterisk 1.8.0 development has been supported by over 500 reporters, more than 300 testers and greater than 200 developers.
You can find a summary of the work involved with the 1.8.0 release in the Asterisk 1.8.0 sumary.

Some of the features that comes with the new Asterisk 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.
For a full list of changes in the current release candidate, please see the ChangeLog:

Jabbering At Cisco! Might Make Us Do Awya With Wispering.

The small company that I work for and part own, uses Cisco gear in every corner and nook. Close to us and in far away lands. And I am sure there got to be more in between.

In front of these myriad networks, we also develop products for prestigious companies. Companies that take most part of your earnings!. Basically we have to convince clients that our products could provide what they require to operate smoothly and more. These clients, they are all over the nation. Some of these, we visit and provide face to face demos. The others we hold teleconferences and provide online demos, using WebEx from, again Cisco. Once we have convinced our clients, we get to setup the products to fit their needs as every one differs in their own way. So we got on with our developers, who expands a few continents. Again it is phones (Asterisk IPPBX) and WebEx. So when we heard that Cisco is definitely going to acquire Jabber, we knew we do not need to whisper, anymore. The other thing is that we have been happily Jabbering on Asterisk with chan_mobile for a while!, ourselves. Our original Jabber server still alive on a Pentium III machine!

So Cisco, we are ready!. Following is the Cisco press release regarding the Jabber acquisition;

SAN JOSE, Calif. - Sept. 19, 2008 - Cisco today announced its intent to acquire privately held Jabber, Inc., a provider of presence and messaging software. Based in Denver, Jabber will work with Cisco to enhance the existing presence and messaging functions of Cisco's Collaboration portfolio.

The acquisition will enable Cisco to embed presence and messaging services "in the network" and provide rich aggregation capabilities to users through both on-premise and on-demand solutions, across multiple platforms including Cisco WebEx® Connect and Cisco Unified Communications.

"Enterprise organizations want an extensible presence and messaging platform that can integrate with business process applications and easily adapt to their changing needs," said Doug Dennerline, Cisco senior vice president, Collaboration Software Group. "With the acquisition of Jabber, we will be able to extend the reach of our current instant messaging service and expand the capabilities of our collaboration platform. Our intention is to be the interoperability benchmark in the collaboration space."

Jabber provides a carrier-grade, best-in-class presence and messaging platform. Jabber's technology leverages open standards to provide a highly scalable architecture that supports the aggregation of presence information across different devices, users and applications. The technology also enables collaboration across many different presence systems such as Microsoft Office Communications Server, IBM Sametime, AOL AIM, Google and Yahoo!. Jabber's platform leads the market in system robustness, scalability, extensibility and global distribution.

The Jabber acquisition exemplifies Cisco's "build, buy and partner" innovation strategy to move quickly into new markets and capture key market transitions. In addition to internal software innovations, Cisco actively employs investments in, and acquisitions of, other companies to support its software strategy; recent purchases include industry leaders WebEx, IronPort, Securent and PostPath.

The transaction will be accounted for in accordance with generally accepted accounting principles. Financial terms of the transaction are undisclosed. The acquisition is subject to various standard closing conditions and is expected to be complete in Cisco's first half of fiscal year 2009. Upon completion of the acquisition, Jabber employees will become part of the Cisco Collaboration Software Group (CSG). CSG is part of the recently established Software Group, consisting of Cisco's major software businesses; including the IOS network operating system, network and service management, Unified Communications solutions, policy management, and SaaS offerings.

Tags: cisco, Teleconference, Web Conferencing, Jabber, Asterisk, chan_mobile, webex

Jabber On Asterisk At Jabber Call Progress Branch

Asterisk Developer SVN server has a new branch now available for some testing, posted Asterisk Developer team member, bweschke. The new branch is at the following SVN URL:
http://svn.digium.com/asterisk/team/bweschke/originate_w_jabber/

This is based on the 1.4 branch and has had the res_jabber module that was in /trunk back-ported to it.

What does it do?

Phase 1 of the work here was to enable call progress messages to be sent via XMPP on Manager Originate actions and also within the Dial application when the appropriate channel variables are set.

What do the messages look like?

I did not get a chance to test it yet but according to the post, they are XML messages. Here is an example message so you get an idea of what the tree looks like.

progress
12345^^CALLID^^12345
PEER/1234567890: channel start

How do I make it work?

For this I have to direct you to post and the poster. It is described well enough to get you going by bweschke at Jabber call progress branch.

Jabber on...

tag: Asterisk, Jabber, XMPP, XML, , IP PBX, CALLID,