Showing posts with label Asterisk. Show all posts
Showing posts with label Asterisk. Show all posts

Wednesday, August 3, 2011

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

Take Asterisk 10 For A Spin WIth PIAF-RED


PIAF-RED with Asterisk10 http://snapvoip.blogspot.com/
For those who are adventures, Tom at PBX in a Flash has put together PIAF 1.7.5.6.3 ISO with the new Asterisk 10 incorporated. informed us.
There were some issues with FLITE and Cepstral TTS but Wardmundy later told us that Cepstral TTS has been fixed by Darren Sessions from Digium.
If you are wondering about Asterisk 10 it is the new versioning system adapted by Digium. What was to be Asterisk 1.10 became Asterisk 10.
You can follow the PIAF-RED discussion at
PIAF RED download sites;

SourceForge - http://sourceforge.net/projects/pbxinaflash/files/
Google - http://nerd.bz/ppelCH
Vitelity - http://nerd.bz/pPegxo

Saturday, July 30, 2011

OpenSIPS 1.6 eBootcamp, Get Ready For A Torrent Of SIP Technology

OpenSIPS 1.6 eBootcamp http://snapvoip.blogspot.com/
OpenSIPS is holding a eBootcamp on OpenSIPS 1.6. This bootcamp is unique in itself because it is held over the Internet for over 7 weeks. Participants will learn from downloading installing and configuring and on to administration of OpenSIPS. It will start on September 19th 2011. The ebootcamp will accept late registrations up until 12th September 2011. So you better hurry. Contact bootcamp@opensips.org for registration and more information
The users will learn how to authenticate users, install a GUI to help with daily administration, forward calls to the PSTN through Dialplan, integrate Asterisk and Voice Mail, Presence agent, Load Balancing, Traverse Nat for SIP and generate CDR records to a Radius Server. At the end, you will learn how to use troubleshooting tools to solve end user problems.
This is basically the same OpenSIPS Bootcamp but help over a broadband connection over a longer period, seven weeks to be exact.

"The live classes will be taken online by web-conference every Tuesday and Thursday 03:00PM GMT, 11:00AM EDT 08:00AM, PDT. To attend this training you will need to have broadband Internet access. You are going to receive a DVD with a virtual machine to run the labs. The virtual machine will be available in the VMWARE format. You can download the free VMWARE player to run the VM. We suggest that you have one separate desktop or server for your VM and at least one IP Phone/ATA in your private labs to complete the training. A LMS (Learning management system) will be available with forums, quizzes and support materials."
The course of learning will cover key objectives;

Install OpenSIPS on a Linux Machine
Routing basics and the default configuration
OpenSIPS authentication using MySQL and Memcache
Install OpenSIPS control Panel.
Connect to the PSTN using Dialplan and Dynamic Routing
Voicemail integration using Call Forward and AVPs
Implement a presence agent
Understand important aspects of load balancing and high availability
Implement SIP NAT traversal using RTPProxy
Account Calls to MySQL
How to use test and monitoring tools to check your configuration

Wednesday, July 20, 2011

AstLinux 0.7.9 Is Yours For The Taking!

AstLinux Firewall
AstLinux 0.7.9 http://snapvoip.blogspot.com/
The AstLinux Team has announced the 0.7.9 release of AstLinux. This update covers several updates and bug fixes related to Asterisk. All current users are encouraged to upgrade at your earliest convenience.
AstLinux is a Linux distribution for Digium's open source IPPBX, Asterisk.

AstLinux contains powerful networking features including:

Supported platforms include:

  • Soekris net4801 and net5501
  • VIA C3 and C7 based systems
  • PCEngines ALIX and WRAP
  • Generic x86 PC Hardware

The following telephony hardware cards are supported:

  • Digium
  • Rhino Equipment
  • Sangoma
  • mISDN v1

Friday, July 1, 2011

Release Candidate 1 of Asterisk 1.8.5 Is Ready For Download.

Asterisk 1.8.5 - RC1 http://snapvoip.blogspot.com/
The Asterisk Development Team just announced the first release candidate of Asterisk 1.8.5 branch and is available for download at Asterisk site, immediately.

The developer team thanks the community for the involving and reporting of issues with the previous release.
The following is a sample of the issues resolved in this release candidate:

  • Fix Deadlock with attended transfer of SIP call Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81,cmaj)
  • Fixes thread blocking issue in the sip TCP/TLS implementation.(Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois, rossbeer, kowalma, Freddi_Fonet)
  • Be more tolerant of what URI we accept for call completion PUBLISH requests.(Closes issue #18946. Reported by GeorgeKonopacki.Patched by mmichelson)
  • Fix a nasty chanspy bug which was causing a channel leak every time a spied on channel made a call.(Closes issue #18742. Reported by jkister. Tested by jcovert, jrose)
  • This patch fixes a bug with MeetMe behavior where the 'P' option for alwaysprompting for a pin is ignored for the first caller.(Closes issue #18070. Reported by mav3rick. Patched by bbryant)
  • Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If he call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup. (Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett)
  • Resolve issue where leaving a voicemail, the MWI message is never sent. The same thing happens when checking a voicemail and marking it as read.(Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard Mudgett)
  • Resolve issue where wait for leader with Music On Hold allows crosstalk between participants. Parenthesis in the wrong position. Regression from issue #14365 when expanding conference flags to use 64 bits. (Closes issue #18418. Reported by MrHanMan. Patched by rmudgett)
  • Fix timerfd locking issue.(Closes ASTERISK-17867, ASTERISK-17415. Patched by kobaz)

For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.5-rc1

Thursday, June 23, 2011

Major Asterisk SCF Build System Changes Introduced.

Asterisk SCF Build System http://snapvoip.blogspot.com/
Yesterday Kevin Fleming, Digium's Director of Software Technologies, posted to Mailing list that he has introduced some major changes to the build system.
Asterisk SCF (Asterisk Scalable Communications Framework) was introduced to the Asterisk community at the Astricon 2010
"On Monday afternoon I merged a significant set of improvements to the Asterisk SCF build system, and along with that there were a number of changes in each of the component repositories to take advantage of the improvements. This message documents the new features (and caveats) of this improved build system:"
There is a long list of the improvements and changes, if you have not subscribed to the mailing list, follow the link below to read the post in it's entirety.
Asterisk SCF Dev

Tuesday, June 21, 2011

The Final Maintenance Release From The 1.4 Branch, Asterisk 1.4.42-rc2 Released

Asterisk 1.4.42-rc2 http://snapvoip.blogspot.com/

The Asterisk Development Team has announced the second release candidate of Asterisk 1.4.42, the final maintenance release from the 1.4 branch. Support for Asterisk 1.4 branch's security related issues will continue for one additional year.(For more information Asterisk support for various branches could be found at Asterisk Versions

The release of Asterisk 1.4.42-rc2 resolves several issues reported by the community and would have not been possible without your participation.

The following is a sample of the issues resolved in this release candidate:

  • Resolve regression with ring groups in the Dial() application (Closes issue ASTERISK-17874. Reported by mspuhler. Patched by elguero)
  • Resolve deadlock when using tab completion on the 'meetme kick' CLI command when an invalid (non-existent) conference room is specified. (Closes issue ASTERISK-17771. Reported, patched by zvision)
  • Resolve issue where voice frames could be dropped when checking for T.38 during early media. (Closes issue ASTERISK-17705. Reported, patched by oej)
  • Resolve issue where DYNAMIC_FEATURES would not activate after a recent DTMF fix. (Closes issue ASTERISK-17914. Reported by vrban. Patched by twilson)

For a full list of changes in this release candidate is available on the ChangeLog:
Asterisk 1.4.42-rc2 is available for immediate download at Asterisk Downloads.

Wednesday, May 25, 2011

Skype (Microsoft) Drops Asterisk Support, Ending Skype For Asterisk

Skype ends Asterisk for Skype Agreement http://snapvoip.blogspot.com/
Within two weeks of acquiring Skype, Microsoft is set out to cut all Open Source ties to Skype, begining with Asterisk, the best open source IPPBX in the world.

According to a letter from Digium to it's customers, Skype will not renew its agreement with Digium that allowed Asterisk to be integrated with the Skype service through Asterisk for Skype, a product by Digium..
"It includes proprietary software from Skype that allows Asterisk to join the Skype network as a native client. Skype has decided not to renew the agreement that permits us to package this proprietary software,” according to the letter.
Skype for Asterisk sales and activations will end on July 26, but Skype has promised to continue supporting and maintaining the software for two more years. Skype may extend this time period “at their discretion.”
But I love what Digium's Danny Windham has to say about the whole issue of Skype;
Is the acquisition good for Skype? Given the valuation, it’s certainly good for the Skype shareholders. But what about the service itself? In a word, no. History suggests that Microsoft’s tendency towards notoriously proprietary tactics will slow the development of Skype as a business tool. Will Microsoft wall-off Skype from competing products, completely? Or, at a bare minimum will Microsoft ensure their products work much better with Skype than those from competitors? Time will tell.

This transaction has attracted speculation from around the globe regarding the future of Skype, Microsoft, Google, Facebook, incumbent phone companies, VoIP, video communications, Apple, the Iphone, Andriod, and I’m sure even pork belly futures. Regardless of your view on how the merger will ultimately shake out, it’s hard to deny at least this one fact: disruptive communications companies and technologies are changing the world at a rapid pace – and in this case, they are generating gigantic value for their shareholders in the process.
Digium Blog

Thursday, March 24, 2011

Keep Your PBX in a Flash ( PIAF ) Running During An Internet Outage!

PBX in a Flash ( PIAF ) Can Run without Internet http://snapvoip.blogspot.com/
If you experienced PIAF server was not able to route any calls after losing your Internet connection, even between local extensions Ward Mundy just Tweeted how we could avoid this.
The solution is to run a Bind DNS Caching Server running on your PIAF box to keep it going with an Internet outage. If you decided to use this solution, and if you are using FDQNs on your phones, the primary DNS server address must internal or external IP address of your server, where your phones are attached.
Ward Mundy, on the PIAF forum, has provided a simple script to run on your server and a set of files to get everything setup properly like ROOT server information and forwarding DNS queries to Google DNS servers.. The script is not shell script so you will have to run each line by themselves. (You can convert it into a shell script, if you have many PIAF servers running.)
After logging into your server with root capability (root or a sudo account) run the following code at the end of the article. Once you run the code and satisfied that everything is fine, run a dig command (e.g. dig inbound1.vitelity.net) for each of the FQDNs of your SIP trunk providers! This will cache the DNS information on your server to be used when the need arises. You can find the info with FreePBX: Tools, Asterisk Info, SIP Info, SIP Registry. Happy calling;

yum -y install bind*

cd /var/named
service named stop
wget http://pbxinaflash.net/source/bind/bind.tgz
tar zxvf bind.tgz
rm bind.tgz

echo nameserver 127.0.0.1 > /etc/resolv.conf

sed -i 's|$nameserver|127.0.0.1|' /sbin/dhclient-script

service named start
service network restart

dig pbxinaflash.com
dig pbxinaflash.com

# look at the output for the second dig command issued above
# 2d dig command should show a very short DNS lookup time, e.g.
#;; Query time: 1 msec
#;; SERVER: 127.0.0.1#53(127.0.0.1)
# IF ALL IS WELL...

#be sure to set named for auto start on bootup

chkconfig  named  on --level 2345
chkconfig --list named

# named should be configured to start with RunLevels 2,3,4,5

Wednesday, March 23, 2011

Are You An Digium / Asterisk Innovator?

http://snapvoip.blogspot.com/
Digium just announced that they are now accepting submissions for the 2011 Asterisk Innovation Awards! The Digium Innovation Award is designed to recognize developers, customers and partners for outstanding achievements with Asterisk that are improving business processes, overcoming technology challenges and enhancing the company’s bottom line. Digium first introduced the Innovation awards at Astricon 2006 and has been an industry wide accepted award and awaited ever since.
This is open to any and all Digium|Asterisk customers and partners world-wide with solutions that are running and in production. There is also no limits to the number of projects and you can submit as many projects as long as you remember that each project requires a separate submission.

Digium will select this year’s winner based upon a number of criteria that include:
  • Your description of what makes your Asterisk-based solution innovative.
  • The amount to which your solution improved processes.
  • The technology challenges that were overcome to achieve your goal.
  • The size or breadth of your solution.
  • The measurable ROI and competitive advantages generated by project.
  • The things that can be achieved with the solution today that couldn’t be previously accomplished.
  • The creativity demonstrated in the implementation of the project
  • The overall presentation of your solution
If you happen to be the winner, you could expect to receive from Digium Asterisk team,

  • Hotel accommodations (2 nights) and airfare for one
  • Presentation of the award with a profile of your company in the AstriCon Conference General session
  • Congratulatory press release from Digium, Inc.
  • Listing on the Digium Web site
  • Chance to highlight the accomplishments of you and your team
  • Recognition by your industry, friends and family
When can applications be submitted? You can submit your projects to the 2011 Digium Innovation Award today! Submissions are being accepted at the following URL:
http://www.asterisk.org/innovation
Submissions are due by August 1, 2011 and winners will be announced at Astricon, 2011.

Octasic’s Vocallo MGW Multi-Core Processor Selected For OpenVox’ DE430P solution

Octasic OpenVox http://snapvoip.blogspot.com/

MONTREAL, Canada and SHENZHEN, China – March 22, 2011 -- Octasic Inc., a leading innovator of media processing and wireless solutions, today announced that its Vocallo MGW multi-core processor has been chosen by OpenVox Communication Co. Ltd, a leading global provider of unified communications equipment and open source Asterisk® telephony hardware and software products. OpenVox has incorporated Octasic’s Vocallo MGW into its DE430P line of PCIe cards for Asterisk® and other open source software, as well as for proprietary PBX, Switch, IVR, and VoIP gateway applications.
The OpenVox DE430P is the latest upgrade to OpenVox’ DE410P solution, which is a four-port T1/E1/J1 PCI card for high-performance voice and data applications. By equipping the DE430P with Octasic’s Vocallo MGW multi-core processor for voice and video over IP, OpenVox can provide its customers with a flexible, low cost solution for superior voice transcoding and offer the ability to add video capabilities as needed. Taking advantage of Vocallo’s patented echo cancellation and voice quality enhancement algorithms, the OpenVox DE430P delivers superior voice quality over T1, E1 and J1 interfaces, enabling users to eliminate echo tails up to 128ms across all 128 channels in E1 mode or 96 channels in T1/J1 modes.
“OpenVox is committed to leading the way in the evolving voice/video applications market, so choosing Octasic’s Vocallo MGW solution was a logical next step for us,” said Lin Miao, president of OpenVox. “With the ability to transcode more than 400 channels from G.711 to G.729AB on a single DSP while consuming less than 2 watts of power, the Vocallo MGW offers amazing performance, allowing us to deliver the highest channel density, best voice quality and lowest power consumption all in one complete solution.”
As the industry’s most complete Media Gateway DSP platform, Octasic’s Vocallo MGW solution enables OpenVox to reduce its system cost by delivering the lowest power per channel, as well as by offering optimal pricing scalability. With a software package that includes wideband audio processing, voice quality enhancement, and video processing for real-time communications, Vocallo MGW provides OpenVox with an optimal solution for today’s latest voice and video applications.
“Octasic’s solutions have been particularly valuable to our customers and partners whose product families are evolving from one application to multiple technologies and applications,” said Fabio Gambacorta, vice president sales and business development at Octasic. “That’s why we’re particularly pleased to have our Vocallo MGW incorporated into OpenVox’s latest enterprise-class solution for voice and video transcoding. This new hybrid card will satisfy their customers’ diversified needs for voice, while also allowing them a smooth migration path for video support.”

Octasic PR Contacts:
Joyce Radnor or Bree Bolognese
SVM Public Relations
+1-617-787-5192 or +1-760-754-7025
joyce.radnor@svmpr.com or bree.bolognese@svmpr.com

Tuesday, February 8, 2011

Running Your Businesses On Open Source Software!

http://snapvoip.blogspot.com/
Danny Windham's, CEO of Digium  speech at ITEXPO East 2011 in Miami about  exciting, real world, large -scale implementations of Open Source software, like Asterisk, MySQL, OpenOffice, Zimbra, Zamanda, SugerCRM and of course Apache!.

TMC

Tuesday, January 25, 2011

Asterisk Exchange, A Quick Q&A With Digium Community Director Bryan Johns

Asterisk Exchange http://snapvoip.blogspot.com/
The Var Guy has conducted what he calls a Quick Q&A, but I think very focused Q&A with Digium Community Director Bryan Johns to extract some facts about the Asterisk Exchange alliance.
Asterisk Exchange has attracted more than 50 partners ranging from hardware providers, ISVs (independent software vendors) and VARs.
At the end of the Q&A session, the Var Guy offers his own views on the Asterisk Exchange. A good read.
The Var Guy

Monday, December 27, 2010

Thursday, October 14, 2010

How To Install And Get The Best Of AGItate.

Installing #AGItate http://snapvoip.blogspot.com/
A few days ago, we wrote about AGItate, "AGItate, Asterisk AGI On Tropo" on taking Asterisk to the cloud. Even though it was not very difficult to understand, I wished for more information on AGItate, Tropo, and Asterisk Interface Gateway (AGI) and my wish was answered today. Actually it was answered a few days ago but only now I found it.

Adhearsion team has posted a screencast series on AGItate;

Get your Asterisk server on the cloud!

Tuesday, September 7, 2010

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.

Saturday, May 8, 2010

Digium® Certifies Panasonic line of SIP Cordless Phone Systems, the KX-TGP500 and KX-TGP550, for Asterisk open source PBX

KX-TGP500 and KX-TGP550, for Asterisk open source PBX http://snapvoip.blogspot.com/
SECAUCUS, NJ--(Marketwire - May 4, 2010) -  Panasonic System Networks Company of America, the leader in business and home telephone systems, today announced Digium®, the Asterisk® Company, has certified the Panasonic line of SIP Cordless Phone Systems, the KX-TGP500 and KX-TGP550, for use with the Asterisk telephony platform. By leveraging Panasonic's market leadership in DECT cordless telephones and Digium's innovative open source PBX, this alliance is a winning combination.
The Panasonic TGP500 series of SIP Cordless Phone Systems boasts outstanding voice quality and a range of productivity-boosting options in a feature-rich desk phone replacement. DECT 6.0 ensures no interference with wireless networks, and the convenient cordless design eliminates the need to run dedicated network wiring to each employee workstation. It is ideal for home and business environments.
Digium is the creator, sponsor and driving force behind Asterisk, the most widely used open source telephony software. The company's product lines include a wide range of software and hardware that enable businesses to implement turnkey unified communications solutions or to design their own voice-over-IP (VoIP) systems. Resellers, telecom professionals and software developers choose Digium's products because only Digium delivers the technical superiority, security and flexibility associated with Asterisk.
"We are excited to work with Digium and their fast growing base of customers using open source, standards-based business phone systems," said Kim Murtagh, Director, Panasonic System Networks Company of America. "As the market leader, Panasonic is committed to delivering innovative features and superior performance."
"The Digium and Panasonic interoperability partnership and certification of IP phones opens new opportunities for Digium channel partners and integrators," said Digium's David Hardy, Director of Business Development. "The Panasonic global footprint and commitment to SIP telephony excellence creates exciting sales opportunities for our partner community and for businesses using Asterisk."
With flexible configuration options, it has never been easier to deploy and expand a SIP-based phone system. Handsets can be easily added wherever needed; all that is required is an electrical outlet nearby. The benefits of SIP are especially compelling in today's business environment, where every dollar counts. The reduced hardware costs and simplicity of routing calls over an Internet connection can add up to huge savings on monthly telephone bills. With the Panasonic TGP500 series SIP phone system, it is quick and easy to add up to 6 cordless handsets -- each with its own number. All that is needed is a single Internet connection and an electrical outlet near the location of the handset. Because it employs DECT technology, the connection is secure and the sound crystal-clear.
TGP500 Series Details

KX-TGP500
The system features a wall-mountable base unit and one cordless handset. It is expandable up to 6 DECT 6.0 cordless handsets and supports up to 8 phone numbers and 3 simultaneous calls. It boasts Wide Band Audio (G.722) and 5 hours Talk Time, 10 days Standby. Its elegant design features a white backlit large LCD on the handset and a Handset locator button on the base unit. It also has a handset speakerphone, 2.5mm headset jack and belt clip.
KX-TGP550
The KX-TGP550 has all the features and benefits of the KX-TGP500 and adds a corded base unit with a large white backlit LCD and 5 hours Talk Time, 10 days Standby, plus a Hands-Free Speaker phone, Handset Call Button on the base unit, and one-touch call transfer with Busy Lamp Indication.

KX-TPA50
The TGP500 systems can be expanded up to a total of 6 cordless handsets by adding the KX-TPA50 cordless handset.
The TGP500 Series will be available in May 2010 through the following Panasonic authorized distribution partners: Jenne Distributors, NETXUSA and D&H Distributing.

Wednesday, May 5, 2010

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.

Thursday, April 22, 2010

Manage Your Hotel Switchboard With Elastix, Easily

Elastix, the other Asterisk http://snapvoip.blogspot.com/
If you are familiar with Elastix, A member of the Elastix community has cooked up a configuration based off PBX In A Flash (PIAF) Hospitality Management System . They have made some minor adjustments make the project work with Elastix.
The Hotel Room Management system for Elastix is a very complete on with abilities like restrict the calls from Ext’s when they are not checked in, per-second billing etc.
Follow the links to get it running on your favorite incarnation of Asterisk.

Elastix Hotel management system could be found here.
If you prefer, the PIAF Hospitality Management System by claudiopizzillo is here.

Friday, January 8, 2010

Fax For Asterisk 1.1.6 Released!


Fax For Asterisk 1.1.6 snapvoip.blogspot.com
Digium has announced the release of Fax For Asterisk 1.1.6 for open source Asterisk. Digium attests that this a a commercial grade FAX add-on module.

The latest release brings number of improvements;

* Support for 64-bit Linux installations.

* Reduction in resource consumption, and increase in performance, of T.38 session handling.

* Simplification of session handling during transition from G.711 to T.38 mode.

* Adoption of the latest Asterisk T.38 negotiation API, ensuring interoperability with a wide range of T.38 endpoints.

The Fax For Asterisk 1.1.6 is could be downloaded from Digium servers (http://www.digium.com/en/docs/FAX/faa-download.php).
Note: Because this release uses the very latest T.38 negotiation mechanism in Asterisk, you need to check the Fax For Asterisk download selector lists for the Asterisk versions supported by the latest release.
Fax For Asterisk  product page.