AsteriskNOW 1.7.1 With Add-on Installer Module For FreePBX Is Ready For You.

AsteriskNOW 1.7.1 http://snapvoip.blogspot.com/
Rod Montgomery recently announced that AsteriskNOW 1.7.1 (Wow, that took a long time to release, nevertheless we are happy!) has been released. This time it has something up the sleeve or trunk, a new module for FreePBX that allows installation of Digium add-on software from within the AsteriskNOW interface. Getting Digium’s G729 codec, Fax for Asterisk, HPEC, or Skype for Asterisk. We’ve also made some changes to make AsteriskNOW even friendlier for newcomers to the Asterisk community.
With AsteriskNOW, you can get among other things;

• VoIP Gateway
• Skype Gateway
• IP PBX
• Call Center ACD
• Conference Bridge

• IVR Server
• Voicemail System
• Call Recorder
• Fax Server
• Speech Server

The DAHDI configuration module and the above modules make Asterisk administration even easier for Asterisk Gurus in a hurry or a newcomer looking to learn. We have been running AsteriskNOW a long time, so much so that this time we even decided to change the machine that it was running, a pentium 4 machine with a newer and speedier AMD Phenom II machine. Like we did,  Download AsteriskNOW run it in a virtual machine or on a dedicated machine. Either way you will enter the future of VoIP IP Telephony, with ease.

FreePBX 2.7 Available For Download.

FreePBX 2.7 snapvoip.blogspot.com
FreePBX team has released the final version of FreePBX 2.7 and available for download from the FreePBX.org. You can also put the FreePBX on the cloud.
The system has gone through numerous changes, addons and bug fixes during it's development phase. The most notable changes were around the Fax handling coming from the new FAX module. The new FAX module for FreePBX has a very intuitive GUI for sending and receiving FAXes through both shared voice/FAX DIDs ( require detection) and dedicated FAX DIDs. Also introduced was the support for FAX For Asterisk (FFA) in addition to the traditional spanddsp based FAX reception.
There are more information regarding the FreePBX 2.7 and a more comprehensive list of features and bugs addressed as well as an overall summary are in the 2.7 Milestone with links and stats to all 142 closed Feature Request and Bug tickets addressed during the developmental run!
Good Job FreePBX team!
FreePBX 2.7 and FreePBX developer releases could be downloaded from official download site.

Put Asterisk On Steroids, Get Elastix!

Elastix Project snapvoip.blogspot.com
We all know about Asterisk and totally grateful for the Digium and the Asterisk community but it is not the easiest system to setup and configure. There have been countless attempts make the deployment of Asterisk easier but one such system, "Elastix" stands out.
Elastix has a great following and the community is very active. Elastix makes it possible to install Asterisk in few minutes and get it running. But that is not all because Elastix includes other related open source projects such as  FreePBX, Postfix, Hylafax, and Openfire and  email, fax, and Jabber instant messaging becomes part of the communications solution for a small business or a solo office.
But the main reason I wrote this is to point you to an article on sourceforge.net blog, where Elastix is hosted. Getting published on the Sourceforge blog is it self a feat as not all 230000 projects get featured on the blog, which is very selective and reserved for the best.
So head over to sourceforge to find out why Elastix put Asterisk on steroids.

Sourceforge Blog
Elastix

FreePBX v3 Developers Preview Announced.

FreePBX v3
Folks at FreePBX has announced the availability of FreePBX v3 Developer preview and invites all who those do testing to iron out the system for rest of the users.
In addition to improving already fantastic PBX system by taking in the user feedback and real world deployement information, FreePBX has fused the TCAPI project and included some contributions from the Bandwidth.com Phonebooth Project. The team has also picked picked Kohana, Doctrine and jQuery as core components to realize the systems design goals.
From the get go, the project has some great goals including but nor limited to;

• Make FreePBX truly telephony engine independent
• A well thought out modular design building on the past 5 years of experience
• Use of a true MVC framework
• Choose best in breed components and build on top of them
• Excellent documentation to encourage developer involvement
• Easily Skinnable

You will find documentation and examples on FreePBX wiki as well as a fully functioning CD, ISO image for download coming soon that will help all of us to get started in evaluating and contributing to FreePBX V3 project!

FreePBX V3 Announcement

Tags: FreePBX, Kohana, Telephony Engine, Phonebooth Project, Doctrine, jQuery, TCAPI project, FreePBX V3

SIP Trunking service From FreePBX.

SIP Service From FreePBX

If you are a user of FreePBX, an Asterisk based IPPBX, the folks at FreePBX have gone one step further and now providing SIP Service for FreePBX and Asterisk based system users.
With many people, SOHO and SMBs going the way of Free IPPBX systems like FreePBX or other Asterisk Based systems, there were SIP Service providers mushrooming everywhere, and sometimes did not stay beyond a life span of a mushroom.
People were always in search of better SIP Service for their business and unlike the Free SIP Service Providers for personal systems, these are business looking for stability and support at a professional levels.
FreePBX, together with long time corporate sponsor of FreePBX, Bandwidth.com has created a new FreePBX branded SIP Trunking service delivered on Bandwidth.com’s network. The goal of this project is to provide seamless interoperability with FreePBX and instant activation of the service so that people will have their telephony service, as soon as they setup their FreePBX Servers..
The teams initial offerings will target the commonly requested Unlimited 2-Way SIP Trunk service complete with one or multiple DIDs on an account and E911 enabled service across the US. This will cost you only, $24.99 a month! DID and Toll free numbers will cost extra but for the cherry on the cake, read the next paragraph.
If you have been paying through your noses for incoming DID numbers, get this, The new service only charges $1 per month per line! So you can get any number of incoming DID numbers at this rate. Toll Free numbers are prices at $2 a month.
We always have been fans of FreePBX and now we have a bunch of other reasons to be so. So if you are looking for cheaper SIP Service with incoming DID lines, follow the link below and register.
FreePBX SIP Service

Tags: Asterisk, FreePBX, SIP Service, SIP Trunking service, Free SIP Accounts

AsteriskNOW 1.5.0 Available Now.

AsteriskNOW 1.5.0

AsteriskNOW 1.5.0, the soft Asterisk Appliance, has been released and available for download and yes we all know it is after a long pause.

If you are an existing user, you will be able to run `yum update` to upgrade your server to current version. It was noted that you might need to run `yum update glibc` in some cases to update the glibc.

Following improvements and additions were added since last beta;
* Updated several packages to latest versions (Asterisk, DAHDI, etc)
* Fixed more permissions issues between Asterisk and httpd/FreePBX.
* Updated to CentOS 5.3

Please use Bug Reports and forums at Digium for reporting bugs, problems and feedback.

Tags: AsteriskNOW, AsteriskNOW 1.5.0, digium, Asterisk, DAHDI, FreePBX, Asterisk Appliance

Google Voice And Asterisk Integration

Googlified Messaging
When you thought Nerd Vittles (Ward Mundy) has surprised you enough, he comes up with another one. This time he is putting ideas into our heads with his recipes for the Asterisk and Google Voice integration. I have already had my GrandCentral number associated with my Asterisk installationS, (yes I have a quite a few Asterisk servers running). I have not been doing much on the Google Voice side of it yet. But after all the Google Voice excitement, subsequent OOMA Plus Google Voice(GVE) and this recipe now, I have no excuse.

If you are looking to get your voicemails trnscribed and emailed (I just get my grandcentral voicemail emailed without transcription) and wring Google Voice and FreePBX (Asterisk) get more out of both, I think Nerds article Googlified Messaging, is a good read.
It is bit technical but easy to follow as most Nerd Vittles article. But if you think you need a laugh after all those technical wrangling you can read some transcribed messages (tests conducted by usesrs like me) you can do it here.

Tags: Google Voice, Asterisk, FreePBX, Ooma, Asterisk, FreePBX, Googlified Messaging, Nerd Vittles

64BIT Elastix 1.5 Beta Ready For Testing

http://snapvoip.blogspot.com/
The elastix core development team and much thanks to Alex Villacis, the first beta version of 64bit Elastix 1.5 have been made available for download. They are getting ready for the first RC but this beta will provide much better Elastix performance in 64 bits OS.

Elastix 1.5 is using Dahdi instead of zaptel and following are the major components of the system;

• FreePBX 2.5.1
• Asterisk 1.4.23.1
• Dahdi 2.1.0.3
• Linux 2.6.18-92.1.22.el5

If you do test the system please submit any bugs that you come across to bug track system.

Tags: Freepbx, IPPBX, VOIP,elastix

FreePBX Time Groups and Time Conditions Explained

After giving you FreePBX 2.5, the FreePBX team continue to give you more! One thing that I really liked was write up on Time Groups and Time Conditions.
FreePBX 2.5 introduced the separate time groups and time conditions modules to ease the time management on FreePBX. If you ever had to deal with "Destination if time matches", you know how hard it is to work backwords, like having a destination before setting up a time condition or having to deal with time sequetialy.
But these modules make the time management much easier and the article explains it very well. Hope I did not already confuse you with those time conditions! Here is the article by Moshe Brevda, FreePBX Development Team.

Update;
Moshe has his own blog and the article could be found at MBrevda.

Tags: FreePBX, asterisk, freepbx 2.5, Time Groups, Time Conditions

FreePBX 2.5.0 Final Released

It’s old news to a lot of you but last two weeks were very busy and missed the release. But following the advise better late than never, I am just informing every one that FreePBX 2.5.0 has gone final!
We tested the FreePBX 2.5 RC3 and were already elated with the results. We will be upgrading our production servers to FreePBX 2.5 soon.
Also you can find all the information on all the modules and who is responsible for them as well as the status of the every portion of the development process could be found here. You can more information and download the release from here.

Thank you FreePBX Team!
FreePBX 2.5 Release

Tags: FreePBX, FreePBX 2.5

FreePBX 2.5 RC3 And OTTS Discounts Offered.

FreePBX has released the FreePBX RC3 for all you testing souls. I think I will attend to it this week end. If you wondered why the FreePBX 2.5 is not released, Philippe explained that lots of features being added, with care being one reason.
The team is also offering OTTS, Open Telephony Training Service at a discount rate, actually a hefty discount of $500. So if you are looking for that training, it is time for you to register. I think I will have a chance to say hello to the team as I plan to attend Astricon and They will also be there. Good luck FreePBX team!
FreePBX 2.5 RC3 release news,
OTTS, Open Telephony Training Service, register here. Do not forget to type LETSGOFINAL for discount code, if it is not typed for you already!.

Tags: FreePBX, OTTS, Asterisk, Astricon

FreePBX 2.5.0 Beta Ready For Testing

FreePBX team has released the FreePBX 2.5.0 Beta with the FreePBX 2.5.0alpha1 available for download.

As mentioned it is still alpha, the feature set is not frozen yet. That means many things to many people but if you are a FreePBX user and if you want to promote any features that is important you that is still open under the 2.5 Milestone. Perhaps you wan to ask for something entirely different from the team. One way to influence them is making a donation.

But if you are just a user you may want to test and provide feedback regarding the features so that the development will get underway for final release.

The Milestone 2.5 link above and the CHANGES document that comes with every new release will show you a summary and detailed list of tickets that have been associated with this release. This is very strongly recommended.

Please note that the OTTS, Open Telephone Training Seminar, has a date change introduced, that we published earlier.

Tags: FreePBX 2.5.0 Beta, FreePBX 2.5.0, FreePBX 2.5.0alpha1, OTTS, Open Telephone Training Seminar

Open Telephone Training Seminar Rescheduled.

FreePBX team has announced a date change on the upcoming Open Telephone Training Seminar. When I published the above article, it was scheduled for August.

The FreePBX team has received numerous requests from possible participants to postpone the date because of vacations and other summer time activities in August. After verifying with those who have already signed up for the event, the team has set the following dates for the training seminar;

Where: Digium Headquarters, Huntsville, AL
When: October 7-10th, 2008

For a limited time, until the End of July, we will also provide an early bird discount of 15%, a $375 savings, but register early to get that discount.

Tags: Open Telephone Training Seminar, OTTS, FreePBX, Digium, IP PBX, PBX, VoIP Training