Blue.Box, Asterisk, FreeSWITCH Configuration Tool For The Cloud.

Blue.Box,, Asterisk, FreeSWITCH Configuration Tool For The Cloud. http://snapvoip.blogspot.com/
Today I came across a new project hosted by 2600Hz, Blue.Box, Blue.box is an open source GUI management tool (administration and configuration) for FreeSWITCH and Asterisk based VoIP systems.
From the information I found, Blue.Box is completely modular and supports multi-tenancy and skinning. Developers will be able to create their own modules and/or add their own features.
Most notable features of the blue.box is it's scalability. Blue.Box can scale from several to thousands of extensions, line, by using one or more severs to handle media and call routing.
The multi-tenant feature enables you to use one blue.box server as several virtual VoIP servers used by multiple entities.

People who will benefit from Blue.Box;

• Hosting providers
• Vendors and system integrators who wish to resell their PBX product.
  
• System administrators who want to have a stable and powerful PBX GUI.
• Developers who want to build modules to upgrade their PBX system or to sell them.

By using it to manage;

• On-premises PBX System
• Hosted or virtualized PBX system
• Video and conferencing system
• Service provider VoIP system

2600Hz Project

mod_pocketsphinx And FreeSwitch Will Help You To Order Pizza.

#mod_pocketsphinx  #FreeSwitch,http://snapvoip.blogspot.com/
A FreeSWITCH app example, Gino's Pizza brings together mod_pocketsphinx, FreeSWITCH (least rev 9194) and a buch of clever thinking and scripts to demonstrate the speech recognizing capabilities of FreeSWITCH.
The page for Example Application for FreeSwitch takes one through every step (very few) to compile and install everything needed to order Pizza through voice lines, be it PSTN or VoIP (Even a free SIP call through a Free SIP account).

PocketSphinx is an open source project by Carnegie Melon University and being used by many people in different ways. Brian West is responsible for mod_pocketsphinx, as I read from the source code file.


Basically the process is very simple four steps if you are knowledgeable in FreeSWITCH but not so hard four steps for people want to order a pizza very badly (there is plenty of help available at FreeSWITCH site to get these four steps acomplished).

1. Please update to at least rev 9194 so this will work correctly. Scoring was changed to be 0 = bad and 100 = good.
2. Build FreeSWITCH™ and enable mod_pocketsphinx
3. FreeSWITCH™ will automatically download and install pocketsphinx
4. enable mod_pocketsphinx in the Modules.conf.xm

Once you have done these then it is getting grammar files ready like;

pizza_yesno.gram

#JSGF V1.0;

/**
  * JSGF Grammar for pizza_size
  */

grammar pizza_yesno;

 = [ yes | yep | correct ];
 =  [ no | nope ];

public  =  ;

Then it is to get your Pizza demo by copying the demo scripts from the source to your working directory.
FreeSWITCH Pizza Order demo
PocketSphinx

VoIP Attacks On The Rise! Secure Your VoIP Servers

#PORT5060 #VoIP.Attacks On The Rise! http://snapvoip.blogspot.com/
ZDNet has an aritcle on rising attacks on VoIP systems. Most of the VoIP attackers seem to employ the method of spamming UDP port 5060. SANs Institute have reported recent spikes in SIP Attacks,) Session Initiation Protocol (SIP) scanning).

We wrote about SIPVcious before, use it to see how strong your systems are.

Even with best security practices, if your clients are using easy to guess passwords, could allow hackers to compromise your system and send your call costs hundreds of thousands of dollars, say during a weekend.
There are many simple little things you could do to limit the damage from these attacks, like changing the SIP ports from the default ones.
Also do not be complaisant just because your server is behind a firewall. Write up some rules for you firewall or like us use some tools like fail2ban to block port scanners. Utilize Blacklist from VoIP Blacklist Project to secure your systems. FreeSWITCH has the script in their source tree.
What ever you do, work a bit hard and keep those hard earned savings in your pocket. We will be posting an article on Asterisk-fail2ban, as soon as Diordna returns from his travels.
ZDNet via  FreeSWITCH

C2 VoIP Patent Fails In Obviousness Test., Says USPTO

#EFF C2 VoIP Patent Fails In Obviousness Test.http://snapvoip.blogspot.com/
In its war against worst patents, EFF has scored a minor but very important victory against one of the top ten worst patents in EFF’s Patent Busting Project,, USPTO has agreed that C2 Patent in question, “Method and Apparatus for Implementing a Computer Network/Internet Telephone System,” does not pass the obviousness test.
This is not a final decision by USPTO as it  needs to wait for C2 response and make an argument in support of it's patent. VoIP Patents
EFF Anouncement via FreeSWITCH

Make Free Calls With GMail Voice Service And FreeSWITCH.

Free Calls With FreeSWITCH And GMail http://snapvoip.blogspot.com/
Folks (actually Anthony Minessale) has cooked up a bit of code to connect FreeSWITCH to GMail Voice interface.
They were able to get multiple calls up to prove the capability. This another confirmation to brilliant design of FreeSWITCH, plug anything you want, almost.
It only requires that you have a GMail account, and FreeSWITCH with mod_dingaling enabled (make sure to enable it load at runtime in modules)
You need to make some changes to  some lines in freeswitch/conf/jingle_profiles/client.xml. Create a Dialplan and you are in business!
The entire code and instructions are available from FS Wiki,
FreeSwitch Wiki

Your VoIP Your Way With Blue.Box At 2600Hz.org

Your VoIP Your Way With BlueBox http://snapvoip.blogspot.com/
I just came across an Open Source VoIP IP Telephony initiative, that look to me very promising. Blue.Box is a configuration and management software utilizing the FreeSWITCH and Asterisk switching libraries. Blue.Box supports multi-tenancy, and supports database and file replication to scale up to thousands of registered devices and simultaneous phone calls. Blue.Box is equally geared to operate in the cloud or on the premise.
Operating under 2600Hz.org, (see below for an explanation of 2600Hz and the connection to the blue.box) the Open Source initiative brings together FreeSWITCH, Asterisk and YATE switching libraries currently centered around blue.box project. They also have other software such as;

Nibble Billing – FreeSWITCH module for simple real-time per-minute billing
Asterisk AMI Class – Simple Interface for Asterisk AMI from anywhere
Package Manager – Karl Anderson’s PHP package management system
Doctrine Polymorphic Modules – Dynamically attach and remove model relations in Doctrine

But it seems they are still in the preparation.
The blue.box is available to download as a ready build ISO, Blue.Box 1.0.1, as RPMs for CentOS, Redhat and Fedora Linux distributions. Source is also available via git.
If you download the ISO be watchful. It is not a live CD, the CentOS based ISO is set up for unattended installation and will format your hard drive and partition automatically before installing the software distribution. You might lose your OS and other software on the computer..
The Blue.Box ISO comes with: CentOS 5.5 x86_64, FreeSWITCH 1.06, FreeSWITCH Sound files, Blue.Box 1.0, Apache 2.2.3, MySQL 5.1.46, and PHP 5.3.2.

The 2600hz and the blue.box project

The 2600Hz and the Blue Box;

2600hz is a frequency that long distance carrier switching equipment formerly used to signal when a long-distance telephone line was not currently in use. It was an important signaling technology in the 60s and 70s. Certain people developed a way to use a device, known as a “blue box”, to generate a 2600 Hz tone on a line that was already in use, tricking the phone company to allow free phone calls to be made by bypassing billing systems when spinning up a new call. More important, however, it was one of the first times that individuals were able to manipulate the telephone system in such a way as to gain direct access to core signaling systems.

Sangoma Releases D100 Voice Transcoding Card And Supports Open Source Telephony Applications

Sangoma D100 http://snapvoip.blogspot.com/
Sangoma has released the D100, a PCI or PCI Express voice transcoding card (converts simultaneous channels of transcoding from one type of codec (e.g. G.711) to another (e.g. G.729)) which is designed for optimum voice quality for low to high density system! Voice transcoding is an essential faction of a VoIP IP Telephony System and the Sangome D100 supports transcoding sessions to up to 30, 60, 120, 240 or 480 channels of any-to-any voice codec conversion per card.
Among numerous capabilities of the card system, the outstanding features include;

    * 30 to 480 transcoding sessions
    * Integrates in Asterisk® and FreeSWITCH™
    * Codecs supported: G.711, G.722, G.722.1, G.726, G.729AB, GSM-FR, GSM-EFR, AMR, AMR-WB (G.722.2), iLBC, L8 (Linear 8K), L16 (Linear 16K)
    * Supports Linux (all versions, releases and distributions from 1.0 up)
    * Compact 2U form factor
    * PCI and PCIe

 Here is the press releasel
TORONTO, ONTARIO--(Marketwire - May 4, 2010) - Sangoma^® Technologies Corporation (TSX VENTURE:STC), a leading supplier of hardware and software enabling server-based voice and data communication applications, today announced the launch of a new series of PCI and PCI Express cards enabling up to 480 channels of voice transcoding per card.
"Voice transcoding is an essential, but costly, component of most VoIP applications and networks," said Frederic Dickey, director of product management at Sangoma. "While software solutions can reasonably handle a handful of channels on a server, our card-based solution allows for hundreds of simultaneous voice transcoding channels in a single server, without compromising on audio quality."
Most IP telephony applications require the use of multiple types of voice codecs, which are used to digitally compress the voice signals at any given deployment. While voice signals from the Public Switched Telephony Network (PSTN) always come in the form of the G.711 codec, the VoIP terminal equipment and networks support a variety of voice codecs including G.729, G.726, AMR, etc. VoIP infrastructure most often needs to include the capability to mediate between endpoints supporting different codecs, but this 'transcoding' digital signal processing task is often costly, resource intensive and can affect the quality of the voice signals if it introduces too much latency or delay.
The D100 card, available in PCI and PCI Express form factors, allows to convert numerous simultaneous channels of voice communication from one type of codec (e.g. G.711) to another (e.g. G.729), without affecting latency or using up precious host CPU resources. The card allows to run 30, 90, 120, 240 or 480 channels of any-to-any voice codec conversion, with unmatched quality.
The D100 software drivers provide "plug-and-play" capabilities for both Asterisk^® and FreeSWITCH™, two leading open source telephony projects. With the compatible drivers, the open source telephony platforms can use the D100 cards as seamless voice transcoding resources. This means that existing Asterisk and FreeSWITCH applications can readily start leveraging the D100 capabilities. Further, the open source telephony software can be used as a gateway or session border controller to provide network-based transcoding services.
In open source telephony software, if a hosted PBX vendor wants to reduce the cost of bandwidth to reach customers' SIP phones via a data connection, the cost, scalability and latency issues of software-based transcoding solutions would be prohibitive. However, the D100 card provides a reliable and cost-effective alternative.
In a contact center operation, the cost of voice transport over a data link to connect remote agents can rapidly become a sizeable part of the operational budget. Instead, the D100 card allows the contact center to use the low bit-rate codecs for connecting remote agents to centralized infrastructure and local telephony lines, thereby reducing the cost of connections to remote agents.
"The D100 card allows us to deliver, as an integral part of our hosted contact center infrastructure, the low bit-rate codecs required by our customers, using a flexible, centralized and affordable open source solution," said Matt Florell, president of the Vicidial Group, an enterprise-class, open source call center suite. "It delivers great audio quality in a small form factor and a very attractive cost per port."
Starting at a list price of US$ 750, the D100 card is currently in customer trials and is expected to be generally available for purchase in the current quarter.

FreeSWITCH Will Get Contineous Support From Sangoma!

Sangoma extends support for FreeSWITCH snapvoip.blogspot.com
Sangoma, a supplier of Voice and data communications hardware supplier bid it's support for FreeSWITCH open source project, the company announced at ITEXPO East 2010.
"Sangoma's contribution has been invaluable since the inception of the movement, both from a financial perspective as well as from a technology partnership level," said Anthony Minessale, lead developer of the FreeSWITCH project, in a statement. "We are excited about Sangoma's continued involvement and active support for the project."
Sangoma is a diamond sponsor, the highest level of sponsorship of the 10^th annual ClueCon Conference (links below). "Since being launched 2006, FreeSWITCH has been growing in popularity, sophistication and stability as a fully-featured, carrier-grade softswitch and telephony platform," said Serge Forest, Vice President of Marketing at Sangoma. "Over the last year, we have seen some of the most innovative and powerful applications being driven by the FreeSWITCH engine.
Following is the complete press release;

ITEXPO EAST 2010, MIAMI, FLORIDA--(Marketwire - Jan. 21, 2010) - As part of its ongoing commitment to the FreeSWITCH open source project, Sangoma® Technologies Corporation (TSX VENTURE:STC), a leading supplier of hardware and software enabling server-based voice and data communication applications, today announced at ITEXPO East 2010 the company is extending its support by becoming a major sponsor of both a leading open source conference and weekly FreeSWITCH community calls. Through its financial and technical support, Sangoma hopes to facilitate the growth of the FreeSWITCH community by increasing the number and types of IP Telephony applications developed.
Sangoma will be a Diamond sponsor, the highest level of sponsorship, of the 10^th annual ClueCon Conference taking place this August 3-5, in Chicago, IL. ClueCon is an annual three-day open source telephony user and developer conference that brings together the entire spectrum of telephony. The conference features presentations by industry leaders on a variety of open source telephony platforms including FreeSWITCH, Asterisk, Callweaver, OpenSIPKamailio, Bayonne and Yate.
Additionally, Sangoma will sponsor the FreeSWITCH community calls, weekly calls where leading developers present topics of interest to the open source telephony community. They also lead discussions on projects, best practices and other topics relevant to open source developers. Sangoma will provide technical support on calls whenever topics related to their expertise are on the agenda.
"Sangoma's contribution has been invaluable since the inception of the movement, both from a financial perspective as well as from a technology partnership level," said Anthony Minessale, lead developer of the FreeSWITCH project. "We are excited about Sangoma's continued involvement and active support for the project."
"Since being launched 2006, FreeSWITCH has been growing in popularity, sophistication and stability as a fully-featured, carrier-grade softswitch and telephony platform," said Serge Forest, Vice President of Marketing at Sangoma. "Over the last year, we have seen some of the most innovative and powerful applications being driven by the FreeSWITCH engine. Its highly scalable and modular software architecture enables developers to reach new heights in the density and flexibility of the solutions they build. Sangoma has leveraged the FreeSWITCH architecture over the last few years to build its products and we are proud to continue our involvement in this highly dynamic IP Telephony community."
Developers interested in Open Source Telephony are invited to join the weekly calls at http://www.freeswitch.org/ and attend the ClueCon conference www.cluecon.com.

Sangoma® Technologies Corporation
http://www.freeswitch.org/
www.cluecon.com.
ITEXPO East 2010

Broadvoice BV16, BV32 Voice Codecs Are Supported In FreeSWITCH Now.

Broadvoice HD Codecs snapvoip.blogspot.com
FreeSwitch has announced the support fro recently open sourced Broadcom's BroadVoice BV16, BV32 Voice Codecs right now.
So if you want to try out the HD voice right away, this is your opportunity. These two codecs were released without any licensing fees or royalties, and under the GNU Lesser General Public License (LGPL), version 2.1. The benefits of these include;

• Low Latency - only 5ms
• Low CPU Usage - typically 1/2 to 1/3 less than comparable ITU codecs
• High Quality - comparable MOS scores to other codecs
• Moderate Bit-Rate - very high coding efficiency

These two codecs joins other high quality codecs, including CELT codec and G.722. added recently.
Read at FreeSwitch

VoIP Security, Asterisk, SIP, Brute Force Attacks, Explained By John Todd Of Digium.

VoIP Security snapvoip.blogspot.com
My thoughts when I read the article saying  “Asterisk attacks are endemic” were ????*&^ and I also knew that John Todd knew what he is doing. So I let the article pass as I was certain it was not the case.
Today Todd has posted an article explaining the misinterpretation of his thoughts and the verbalization of the same.

My comment in the article was not that “Asterisk attacks are endemic”, but that SIP-based brute force attacks are endemic.  Every SIP system that is open to the “public” Internet is seeing large numbers of brute-force attacks.  Sites that have weak username and weak password control will be compromised – this is little different than email accounts being taken over by password-guessing systems and used for sending floods of email.  The significant difference is that when someone takes over a SIP platform to make outbound calls, there is usually a direct monetary cost, which gets people’s attention very quickly.

We all know that Asterisk is used world over and is the favorite SIP based telephony platform is likely to attract people who are interested in hacking or attacking the system. But we also have seen security measures taken by Asterisk as well. Just like the one released day before yesterday, AST-2009-07, where the advisory and the fix was released simultaneously.
But what Todd wrote makes sense and as Asterisk, Broadsoft, Cisco, Kamailio, OpenSER, FreeSwitch, Avaya are all vulnerable to brute force attacks and every bit of  information will help. Whether it is news worthy or not.
Asterisk and SIP Security Redux

snom And Cudatel Partner To Provide Best Of OSS IP Telephony

snom partners with Cudatel snapvoip.blogspot.com

snom making partnerships with eqipment makers to provide end to end VoIP IP Telephony solutions to small to medium sized business'. The latest is their partnership with Cudatel,  a wholly owned subsidiary of content security vendor Barracuda Networks Inc. snom is know world over for developing and manufacturing voice over IP (VoIP) phones for enterprise and residential markets.

The partnership paves the way for businesses to deploy snom's portfolio of Linux-based SIP VoIP phones with CudaTel's FreeSwitch based IP PBX and communications appliance, CudaTel Communication Server. FreeSWITCH is a free open source software communications platform for voice and chat-driven products released under the MPL free software license.
snom and Cudatel together have conducted rigorous interoperability and certification testing to ensure the joint solution is fully integrated and deployment ready and that the two companies' gear will integrate seamlessly and ease VoIP deployments.

We are pleased to be working with CudaTel as we continue to partner with the best-in-class technology partners, CudaTel is taking the expertise Barracuda Networks established with its data communications infrastructure product portfolio and bringing this same innovation into open source VoIP with a robust, FreeSWITCH-based platform. Teaming up with CudaTel creates a full, end-to-end VoIP solution for companies of any size built on an open source model, said Dr. Michael Knieling, Executive Vice President of Marketing and Sales for snom Technology AG.

 The CudaTel Communication Server is offered in four models with prices starting at $2,000. The IP PBX are capable of supporting 25 to thousands of VoIP telephones with a range of advanced services. These advanced services include automated handset provisioning and deployment call routing, call recording, call queuing, parking, unified messaging and voice mail, conferencing and conference bridges. The  CudaTel Communication Server is also capable of providing automated attendant services, Active Directory integration and follow-me functionality. CudaTel also offers unlimited 24x7 phone and email support with each CudaTel server and does not require per-user, per-box or per- feature licensing fees.

 CudaTel was built from the ground up to be an innovative and affordable VoIP system and designed to work exclusively with best-of-breed SIP phones like the endpoints offered by snom, Partnering with snom ensures that our CudaTel Communication Server customers will have access to telephones that are easily deployed and provisioned, and that offer a host of capabilities and functions. said Sean Heiney, CudaTel product manager and director of new product initiatives for Barracuda Networks.

 With the new partnership, expect to use  snom's portfolio of SIP-based commercial-grade VoIP phones, including the snom 320, snom 360 and snom 380, as well as the new snom 820 and the snom 870 color touch-screen phone with Cudatel eployments, existing or planned. Also supported are the snom MeetingPoint conferencing phone and the snom m3 IP DECT phone from snom. snom was the first company to provide "zero touch" IP Phone deployment by following the TR-069 standard.
snom Technology AG.
Cudatel

Cudatel, FreeSwitch Based IP PBX System Debuts.

CudaTel, FreeSwitch On Baracuda Systems.

Cudatel a division of Barracuda Networks announced a line of FreeSWITCH based IP PBX systems also named CudaTel.

According to their website, CudaTel Communication Server does not have per-user or per-feature fees, eliminating guess work out of budgeting. Among many attractive features, automated phone deployment (Systems carry built in phone provisioning for Polycom and Snom phones), Active Directory integration, and a simple Web-based user interface stand out.

Currently Cudatel has four systems supporting 10-250 concurrent calls. Systems start at $1999.

CudaTel features:

• Automatic Phone Provisioning
• Call Conferencing
• Group Calling
• SIP phone and provider support
• Automated Attendant (IVR)
• Native phone line hardware integration
• Call monitoring, barging, and bridging
• Web-based performance and usage statistics
• Web-based administrative interface
• High definition voice quality
• Call recording
• Call routing
• Scheduled call routing
• Follow-me
• Call parking
• Active Directory and LDAP integration
• Local Users, Groups, and Policy
• Automatic Extension and phone number (DID) assignment
• Multi-site integration
• Encrypted VOIP support

Cudatel

Tags: freeswitch, Cudatel, Call Conferencing, free sip phone, Encrypted VOIP, Automated Attendant, IVR

AstLinux 0.6.7 Ready For Download

AstLinux Distribution
Kristian Kielhofner has released a newer version of AstLunux distribution, AstLinux 0.6.7. He has made a quite a few changes and those are available from his site.
AstLinux is a custom Linux distribution built around Asterisk, the Open Source IP PBX. But you will see more than Asterisk with AstLinux. Many VoIP applications, many of them, my favorites, like OpenSIPS and FreeSWITCH are also made available from the project.
AstLinux is an ideal choice for embedded and / or commercial Asterisk based solutions. I have been running an AstLinux server for a while now.
If you already have AstLinux, you can upgrade via upgrade function in the web interface. You can also do the same from the command line:
"upgrade-run-image upgrade http://mirror.astlinux.org" sans quotes. If you are new, download the complete ISO or the img.
More information available from AstLinux site.

Tags: ASTLinux, OpenSIPS, FreeSWITCH, Asterisk

FreeSwitch 1.0.2 Released

http://snapvoip.blogspot.com/


The FreeSwitch team has released a new version of FreeSwitch to celebrate the new year. The release comes with a bunch of new fixes and features. Most notables are the benefits of teaming with Polycom. FreeSWITCH team has made it extremely easy and simple to reach into Polycom's wideband (WB) and ultra-wideband (UWB) Siren(tm) family of codecs. This is a very important milestone as many commercial, SMB and SMEs use polycom equipment within their networks.

Another exciting development which we wrote about earlier (48k CD Quality VoIP with CELT), is the addition of the 100% open-source CELT codec. CELT supports sampling rates higher than the 44.1kHz rate used by traditional CDs.
To prove the value of the CELT codec, FreeSwitch members Anthony and Brian played/listened to a guitar solo over a 48kHz session via FreeSwitch and Celt. The listener, Brian has summed it up as awsome!.
These are very valuable codecs to VoIP Users and VoIP Developers. About 6 years ago, I helped to develop a VoIP Telephony system based H323 Gatekeeper and G729 codec. The licence fee for G729 almost stopped the project in its tracks.
With these codecs people will be able to conduct HD quality conferences over VoIP. Whether over private or public networks, the use of WB Telephony, UWB Telephony, and HD telephony will reach very high denominations, provided that you use technologies like FreeSwitch and CELT.
There is more. Another two most notable features added are the FreeSwitch fax capability and a FreeSWITCH remote client. Thanks to the fax capabilities, dialplans now can be used to route inbound or outbound fax calls.
The Remote client is capable of connecting to local or remote instances of FreeSWITCH. An INI-style configuration file allows for multiple profiles. Think Administrates handling multiple FreeSwitch instances!
There are many more items to list and the changelog list 290 of such items. So read this article about the release and then install the new version!
Thank you FreeSwitch Team!

Tags: freeswitch, CELT codec, G729, Polycom, FreeSwitch fax, FreeSWITCH remote client, VoIP Conference, HD VoIP, WB Telephony, UWB Telephony, HD telephony, H323 Gatekeeper

48k CD Quality VoIP with CELT, Another First From FreeSwitch!

http://snapvoip.blogspot.com/


FreeSwitch has started supporting CELT, a new open source audio codec that allows for CD-quality voice and sound transmission. This certainly a boost for VoIP World. CELT is an "ultra-low delay" audio codec that supports both voice and music and said to carry 48kHz VoIP in 48kbps of bandwidth. Compare that to PCM mu-law codec, G.711u, which consumes 64kbps of bandwidth to carry 8kHz audio!
I have not yet tested the codec but plan to integrate the mod_celt and the new FreeSwitch client very soon in my testing rig. Main while, you can find more information at FreeSwitch Post and the following links.

For more information on CELT please visit the project home page and for more information on other free codecs please visit the Xiph page.

Tags: freeswitch, PCM mu-law codec, G.711u, CELT codec, open source audio codec, FreeSwitch client, Xiph

The Open Source Ruby Framework Adhearsion

http://snapvoip.blogspot.com/
The open-source Ruby framework Adhearsion (What's Not Telling About Asterisk? ) has updated the roadmap and it is very promising. The day that everyone will be able to make their own VoIP application or platform is coming near. For me the configuration file generators with Asterisk has been the high light. But due to time costraints, testing using has been a few jabs rather than actual productive work. The roadmap.
But the roadmap is entising me to pay more attension to the development process and some more added benefits are coming in the form of FreeSWITCH compatibility and YATE Compatibility.
If you are interested, this tutorial might be of help.

Tags: Adhearsion, Asterisk, freeswitch, yate, Open Source Ruby Framework

mod_fax, Fax Support For FreeSWITCH In The Works

I made a visit to FreeSWITCH (I made sure I typed FreeSWITCH) is working on providing fax support via spandsp. The module is mod_fax. The code is already checked in and of course you can start beta testing. The ever present and busy Anthm released the news.

if you did not know about FreeSWITCH, it is an open source telephony platform that you can used to create any product that consists of voice and chat driven products ranging from a soft-phone up to a soft-switch. It has been used as a simple switching engine, a PBX, a media gateway, and as a media server to host IVR applications using simple scripts or XML to control the callflow. In my configuration, FreeSwitch is providing Media Gateway services.

Tags: freeswitch, Asterisk, spandsp, mod_fax, switching engine, a PBX, a media gateway