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

Wednesday, February 1, 2012

Digium Phones, Worlds First Asterisk Phones, Introduced At ITEXPO 2012!


Digium Phones, Worlds First Asterisk Phones, Introduced At ITEXPO 2012! http://snapvoip.blogspot.com/

Digium in a surprise move introduced it's own phones at the ITEXPO 2012, the phones;
"* D40—An entry-level HD IP phone with 2-line keys. This is Digium’s best value phone, designed for any employee in the company.
* D50—A mid-level HD IP phone with 4-line keys and 10 rapid dial/busy lamp field (BLF) keys with an easy to print paper label strip for the user’s most important contacts. This model is perfect for users who spend a lot of time on the phone.
* D70—An executive-level HD IP phone with 6-line keys and 10 rapid dial/busy lamp field (BLF) keys and real-time status information displayed on an additional LCD screen, allowing users to quickly navigate through up to 100 of their most important contacts. Designed for administrators or executives, the D70 offers top-of-the-line features."

Digium plans to have general availability of these new phones in April 2012. The MSRP for these models is as follows: D70 - $279, D50 - $179 and the D40 - $129. For more information, visit http://www.digium.com/phones.

This is great news to anyone who is interested in Asterisk and Digium, meaning open source telecommunication or VoIP IP Telephony as a whole.

Press Release;

Digium® Introduces World's First Phones Designed for Asterisk®

Simple configuration and game-changing app engine set new bar for IP phones

ITEXPO, Booth 901 (Miami)-February 1, 2012-Digium®, Inc., the Asterisk® Company, today introduced a new family of high-definition IP phones. They are the first that are engineered to fully leverage the power of Asterisk, the world's most widely adopted open source communications software, and Switchvox, Digium's award-winning unified communications (UC) system. With Digium technology on both the server and the phone, users will benefit from the best possible performance, unprecedented integration and a uniquely customizable phone system.

"Digium's new phones mark the launch of the next chapter in our history of innovation," said Danny Windham, president and CEO of Digium. "These are the first phones designed specifically for Asterisk-with the tightest integration possible between the phones and Asterisk. The success of Asterisk began with the transfer of power from the hands of the proprietary phone vendors to the hands of end users and administrators of phone systems. And now we've done it again by bringing control to the desk phone. These phones are absolutely the easiest to install, integrate, provision and use with both Asterisk and Switchvox. And best of all, we've done all of this at a very competitive price point, providing our customers with the best value in business phone systems."

Asterisk has always been about flexibility, allowing integrators and developers to create highly customized solutions. Likewise, Digium phones include an app engine with a simple yet powerful JavaScript API that lets programmers create custom apps that run on the phones. They aren't simply XML pages; Digium phone apps can interface directly with core phone features.

"The app engine is a game-changing feature that will allow developers to write their own apps that run on the phones," explained Mark Spencer, founder and chief technology officer for Digium. "We have a community of more than 80,000 users and developers who create amazing things with Asterisk. I look forward to seeing the cool apps they will create with these innovative phones. As usual, we're enabling developers to create solutions limited only by their imaginations."

Digium has leveraged this unique programming interface of the phones to create a suite of productivity applications that work with both Asterisk and Switchvox. Switchvox includes a unique web interface called Switchboard that gives each system user control of their personal communications environment. Digium has extended the capabilities of the Switchboard to the phone, putting advanced features like presence management, searchable contact directory, queue monitoring, recording and voicemail control, all at the user's fingertips.

"We've had the opportunity to work with the phones early on, and for the first time, we're using phones truly designed for Asterisk," said Chris Green of Bema Information Technology. "Digium has delivered phones that simplify the most time-consuming portion of a system deployment-installation and provisioning. The phones automatically locate the server and assign the correct users, making our deployments faster than ever before. Additionally, the tight integration with Switchvox offers our customers an awesome user experience that's just not possible with other phones."

The Digium phones include the following models:

  • D40-An entry-level HD IP phone with 2-line keys. This is Digium's best value phone, designed for any employee in the company.
  • D50-A mid-level HD IP phone with 4-line keys and 10 rapid dial/busy lamp field (BLF) keys with an easy to print paper label strip for the user's most important contacts. This model is perfect for users who spend a lot of time on the phone.
  • D70-An executive-level HD IP phone with 6-line keys and 10 rapid dial/busy lamp field (BLF) keys and real-time status information displayed on an additional LCD screen, allowing users to quickly navigate through up to 100 of their most important contacts. Designed for administrators or executives, the D70 offers top-of-the-line features.

Digium plans to have general availability of these new phones in April 2012. The MSRP for these models is as follows: D70 - $279, D50 - $179 and the D40 - $129. For more information, visit http://www.digium.com/phones.

ITEXPO 2012 attendees are encouraged to visit Digium at booth 901 to preview the phones.

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