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June 05, 2005

astGUIclient version released 1.1.1



Hello,

We've released another update to our Asterisk GUI Client suite: 1.1.1

http://astguiclient.sf.net/

Screen shots:
http://astguiclient.sourceforge.net/screenshots.html

The client suite runs on both Windows and UNIX, includes the VICIDIAL auto-dialer and is free as in GPL. (the suite is not an asterisk configuration tool) This package is geared towards Asterisk installations with SIP,IAX or Zap phones and Zaptel or IAX trunks.

In addition to many bug fixes and new features, we've added a web-based version of the astGUIclient that only requires a web browser to use and allows you to extend the functionality of your phone wherever you are.

Here are some of the features of the astGUIclient web client:
-Grabs live call info from a DB updated every second
-Displays live status of users phones and Zap/IAX/SIP/Local channels
-Allows calls to be placed from GUI and directed to phone
-Allows intrasystem calls at the click of a button
-Allows call recording at the click of a button
-Allows conference calling through GUI
-Administrative Hangup of any live Zap/IAX/SIP/Local channel
-Administrative Hijack of any live Zap/IAX/SIP/Local channel
-Administrative switch user function
-Call Parking sends calls to park ext and then redirects to phone ext
-CallerID popup with links to open custom web pages
-Voicemail display and button to go right to check voicemail
-Allows Blind transfers of calls to specific voicemail boxes
-Allows Blind transfers of calls to intrasystem extensions
-Allows Blind transfers of calls to external numbers
-Send to Voicemail directly from the inbound call popup window without answering
-All client phone connections are shown not just the first
-Allows transfers to conferences
-Call parking with callerID
 
To upgrade from 1.1.0 you need to update all of the server apps and web pages and run the update sql script.

Let me know what you think.

Thanks,
-Matt

Posted by Dal at 12:46 AM | Posted to Asterisk Software

IAX Phone Pro - Open Beta Test

IAX Phone Pro - Open Beta Test

Sokol & Associates is pleased to announce an open beta test of the new IAX Phone Pro. IAX Phone Pro is a professional grade IAX soft phone designed for use with the Asterisk open source PBX.

Among the new and improved features:
-Dial/answer/hold/recall/reject
-Multi-number advanced speed dial.
-Standard and innovative "Tool Bar" skins.
-Handles "iax:", "sip:" and "tel:" URLs
-Integrated web browser for "co-browsing"
-Integrated call recording and playback.
-Advanced phone book with CSV import.
-Advanced call log with CSV export.
-Speaker Ph-one
-Audio mute.
-Auto answer.
-Intercom calling with password.
-Multi-server registration.
-Audio Codecs: uLaw, aLaw, GSM, iLBC, Speex.
-Server-by-server codec setting.
-Call statistics.
-Local or server-side call forwarding.
-Local or server-side do-not-disturb.
-TAPI integration for Outlook, ACT, Goldmine, etc.
-Direct IP to IP calling
-Dial by IAX or SIP URI (URL)
 
This is a time-limited version of the full product. Some features which are still under development have been disabled. Updates will be provided as those features are implemented. Please download the phone and let us know what you think.

IAX Phone currently supports the Microsoft Windows family of operating systems. A Mac OS X system may be developed depending on interest.

DOWNLOAD IAX PHONE PRO:
http://www.astricon.net/phone/ipbeta.php

To see a live demo of IAX Phone Pro, drop by the Ipsando booth at AstriCon Europe!


Steven Sokol

Posted by Dal at 12:39 AM | Posted to Asterisk Software

FCC Puts VoIP E911 Laws Online

The FCC in the USA has put online information on VoIP providers being required to provide E911 services: (PDF Format)

Commission Requires Interconnected VoIP Providers to Provide Enhanced 911 Service.

Order

News Release

Martin Statement

Abernathy Statement

Copps Statement

Adelstein Statement

Recorded Audio/Video Webcast of Commission Meeting

 

Posted by Dal at 12:27 AM | Posted to VoIP Politics

June 02, 2005

1.0.8 Release Candidate

Hello everyone!

It has been a while since the 1.0.7 release, and I have fixed a lot of stuff since then. I think it's about time to make another release.

I realize that there are still some outstanding issues, but it's nearly impossible to bring that down to zero. However, I'm open to discussion on anything that someone may feel is a "show-stopper".

I am on IRC as "drumkilla" and also available by email if anyone has any questions or comments.

Please test and report any issues on the Asterisk issue tracker, even if it is just a note saying that you have no problems at all! I will release 1.0.8 once I have had enough reports of people successfully running the latest code from the v1-0 branch of CVS.

Thanks!

Russell Bryant

UPDATE:

I forgot to mention that for bug reports/success stories, please post to the following issue on Mantis.

http://bugs.digium.com/view.php?id=4424

Russell

Posted by Dal at 09:19 PM | Posted to Asterisk Releases

How to: VoiceBlue VoIP GSM Gateway with Asterisk IP PBX

Excerpt: Main scenario

Suppose we have an IP network to which an Asterisk IP PBX, several SIP telephones and an Ateus VoiceBlue GSM gateway are connected. This typical configuration is shown in the figure below. Furthermore, suppose that the network is addressed as shown in the figure and GSM numbers are all numbers starting with 6,7,8 and containing 9 digits. For configuration simplicity, use SIM cards from one GSM provider.

Now say that all incoming calls are answered by the gateway, which replays the invitation message and waits for 10s for another DTMF dialling. After this timeout, the gateway dials extension 111, which is a dial-in to the operator.

Full Article: Click Here

 

Posted by Dal at 12:21 AM | Posted to Asterisk Case Studies

June 01, 2005

Please support the Zeroconf project - Spread the word

The Astmasters Zeroconf Project has posted asking for people to help spread the word on the project:

Maybe it was Memorial Day, or maybe it's because the French rejected the EU constitution. Whatever it was, we are disappointed that almost nobody picked up on the news that we've added Bonjour/Zeroconf support to Asterisk, SineApps and SIP Center being the notable exceptions reporting it.

The marriage of Zeroconf with open standard VoIP protocols is the biggest news in VoIP so far this year. Seriously!

Yet both the traditional media and the blogging scene appear to be in a frenzy talking about silly game consoles this week, stuff which represents absolutely no value whatsoever to the human species, and by doing so they only aid the marketing efforts of mega corporations who could easily pay for the advertising.

Our project, which solves a very serious problem of VoIP today, could really use a little more promotion and we deserve it, too. And here is why ...

Just about any VoIP device and service out there is a total disaster when it comes to setting it up and making it work. Yet, people are used to plugging in a telephone set to the outlet in the wall knowing that it will just work. VoIP will have to deliver the same ease of use or it will be doomed to failure. The only two systems which have reached that level are iChat and Skype.

And by the way, both iChat and Skype use Bonjour/Zeroconf.

Do those analysts and bloggers not realize that adding Zeroconf to systems like Asterisk and user agents like SFLphone will make true plug-n-play-works-out-of-the-box VoIP devices and services possible for non-proprietary open standard based systems as well?
 
We need more developers to start other projects, we need more mind share, we need sponsorships. To get there we need publicity. We make free software, so it's only fair enough that we should get free publicity in return. You get to use free software, so it's probably only fair enough that we ask for your help to claim free publicity to further our cause.

So please spread the word, bother the people you know could be of help. Bother the Mac sites, the VoIP sites, the news sites. They may ignore three or four or five mail messages, but they won't ignore 30, 40 or 50. There are about 250 people on this list right now, if everybody sends one message to a news editor nearby, it should have an effect.

Mind you, Henry Sinnreich, who is the Linus Torvalds of SIP, found our project interesting enough to kindly respond to an email we had sent him - not that we expected any response - god bless him! If he finds what we're doing is interesting enough to be worth his valuable time, then perhaps so should other, lesser important people in the VoIP/SIP world ;-)

thank you for your patronage and support
 

Posted by Dal at 11:56 PM | Posted to Asterisk News

Asterisk@Home 1.1b has been released



Asterisk@Home has been updated to include SugarCRM:

We have replaced the simple contact management system in Asterisk@Home with SugarCRM a full CRM system. This might seem like over kill for a home PBX but Sugar has the best contact management we have seen. With click to dial functionality and the ability to import data from other contact managers it’s a great fit for Asterisk@Home.

We have also added new version of the usual Asterisk software AMP and Flash operator panel.

Download:
 

Posted by Dal at 11:51 PM | Posted to Asterisk Software

AsterTest - The Asterisk Performance Project



The Asterisk Performance Project run by Joachim Vanheuverzwijn (zoa) aims to understand the performance and scalabilty one can expect from their Asterisk system. AsterTest attempts to make sense of what kinds of factors affect Asterisk performance including protocols, hardware and kernel configurations. The site contains a forum for the discussion of Asterisk performance testing and scaling, plus the PowerPoint presentation Joachim and Davy Van De Moere presented at Astricon Atlanta 2004.

The sites mission objectives include:
- measure & improve the codec performance, by finetuning compiler flags & using assembly optimizations.
- improve the complexity of the iax2 protocol.
- get rid or increase any hard limits found.
- provide compatibility information for all protocols and a lot of hardware out there.
- increase scalability & failover
- provide a standardized measurement / load generating / stress testing tool for asterisk.
- give information on how to convert audio formats to save on translation paths.
- anything you suggest....

The AsterTest site can be found at:
 

Posted by Dal at 12:13 AM | Posted to Asterisk Software

Asterisk Management Portal Updated



The web-based Asterisk administrator GUI, called Asterisk Management Portal, or AMP, has been updated to v.1.10.008. New features includescheduled backup and restores, automatic call recording, the ability to use any type of trunk (before it was limited to Zaptel), and other enhancements.

AMP is an open-source projectsponsored by the Calgary-based Coalescent Systems. It integrates MySQL, Perl, PHP, Apache, and Asterisk into a uniform, cohesive PBX system.
 
Visit http://amp.coalescentsystems.ca to check out the details.
 

Posted by Dal at 12:09 AM | Posted to Asterisk Software

Asterisk pbx on wrt54g Wireless Access Point




the linksys wrt54g has been the target of many hacks taking it far beyond its original intended mode of operation; now going so far as becoming a personal telephone company. David DeLauro has set up an asterisk pbx system on top of the the openwrt firmware package. asterisk is a fully featured pbx system. in theory this set up could allow you to wire your whole house through one voip connection. you can even script the dialing sequence so that certain prefixes are handled by different carriers. the router has limited storage space, but you could probably use the voicemail features by mounting network storage. with enough ingenuity you should be able to pipe all of your communications and entertainment over ethernet.

Click Here for more Info 

Source: Gregarius.net - hack a day
Editor Note: I found this little tidbit. Being a wi-fi junkie also, really would like any additional info about this hack.

Posted by Dal at 12:03 AM | Posted to WiFi / Wireless

Canadian firm scores with open source call center

SearchEnterpriseLinux.com has published an article about Aheeva and Asterisk.

Excerpt:
"The decision to create a call center based on open source technology turned out to be a profitable one for Aheeva Technologies.

The Montreal-based firm had long specialized in assisting firms choose, develop and manage interactive contact center systems based on proprietary technologies. But when Aheeva began moving forward with plans to branch out and offer hosted call center services, it found that the open source private branch exchange (PBX) Asterisk made the most financial and technical sense."

 

Posted by Dal at 12:00 AM | Posted to Asterisk Case Studies

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