Hear the airport's control tower in Seville (TWR LEZL) . This code is ready to collect real-time audio streaming from a source on the Internet where he leaves his radio tower broadcasts, and proof of concept allows us to see some of the opportunities the system without having a radio in our system . In this way, we can assign the background music "radio" for when a call for a DID, an extension or a queue of agents.
First, edit the file musiconhold_custom.conf and add the following:
The Asterisk Development Team has announced the release of libpri version 1.4.12-beta1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/
This beta release contains some fixes and several new features, among them:
1.) ETSI and Q.SIG Call Completion Supplementary Service (CCSS) support
2.) ETSI Advice Of Charge (AOC) support
3.) ETSI Explicit Call Transfer (ECT) support
4.) ETSI Call Waiting support for ISDN phones
5.) ETSI Malicious Call ID support
Fluency ACD - Automatic Call Distribution delivers a powerful solution for all Customer Contact Management with two guiding principles, enhanced Productivity and complete business accountability.
1] Enhanced Productivity- To ensure more customers are professionally dealt with by the best qualified people within the organization, maximizing staff talk times during an average day and automating much of the business process allowing personnel to focus on more productive tasks
2] Complete accountability- No other software within a company department provides such complete Management information in both a Live and Historical manor to allow companies to make informed decisions on the Service levels they are aiming to deliver and the staff head count required in achieving this goal.
QueueMetrics version 1.6.1 has been released. This release offers a large number of bug fixes, misc improvements and new developments. Some are worth pointing out:
* Hotdesking - a lot of our customers complain that after switching to ADDMEMBER-style logins, it is not possible to keep agents separate from their extensions. Hotdesking solves this in an easy and efficient manner, and makes it all transparent if you use the Agent's page.
* To make implementing Hotdesking easier, we release the new User Manual with a section explaining all the details, and we also release version 4.0 of the TrixBox/FreePBX tutorial using hotdesking.
VOIP Today magazine releases its 10th issue the postscript to www.voiptoday.org .
VOIP Today magazine is a freely available and independent online publication presenting up-to-date VoIP news and information covering all aspects of the VoIP technology, internet telephony solutions, networks, phones, security, internet telephony marketplace, mobile communications, VoIP forums and call center solutions. It has strong relationships with members of the VoIP community and is rapidly building a unique, high-quality community of VoIP users and vendors.
VoIP Today magazine is building tomorrow's VoIP community.
Share in building tomorrow‘s community by joining VOIP Today community
The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta1.
This release marks the beginning of the testing process for the eventual release of Asterisk 1.8.0.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, http://issues.asterisk.org/. It is also very useful to hear successful test reports. Please post those to the asterisk-dev mailing list.
Asterisk 1.8 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. For more information about support time lines for Asterisk releases, see the Asterisk versions page.
Pyst consists of a set of interfaces and libraries to allow programming of Asterisk from python. The library currently supports AGI, AMI, and the parsing of Asterisk configuration files. The library also includes debugging facilities for AGI.
Download from Sourceforge project page.
Yealink and Elastix signed the cooperation agreement on making Yealink the exclusive official phone brand for Elastix Certified Engineer Training (ECE Training), which indicates the strategic partnership built between Yealink and Elastix. This close cooperation guarantees the excellent training product kit and solutions for trainees.
“Elastix is a well-known IP Telephony and Unified Communications Solution. It is a great honor for Yealink to be the exclusive phone brand used in the Elastix ECE training. This cooperation is a strategic decision made between Yealink and Elastix.
Yealink and Elastix signed the cooperation agreement on making Yealink the exclusive official phone brand for Elastix Certified Engineer Training (ECE Training), which indicates the strategic partnership built between Yealink and Elastix. This close cooperation guarantees the excellent training product kit and solutions for trainees.
“Elastix is a well-known IP Telephony and Unified Communications Solution. It is a great honor for Yealink to be the exclusive phone brand used in the Elastix ECE training. This cooperation is a strategic decision made between Yealink and Elastix.
One single platform to manage all your network services! eBox is a Network Gateway, a Unified Threat Manager, an Office Server, an Infrastructure Manager and a Unified Communications Server.
eBox Platform is the Linux Small Business Server that can act as a Gateway, Unified Threat Manager (UTM), Office Server, Infrastructure Manager, Unified Communications Server or a combination of them. All network services are based on the same technology and fully integrated!
eBox Unified Communications
* eBox Unified Communications Server manages all your communications. Allows easy configuration of: User management, Email, Instant messaging and VoIP.
[posted by by Ruben on open-voip.com]When I took the first look at Freesentral I immediately saw the fresh approach to Open Source PBX configuration. It's fresh due to not being bloated like a few other contenders out there.
Introduction to what's in a PBXIf we take away all the whistles and bells and bloated featurism found in nearly all PBXes today, both open source and closed, commercial, offeirngs, we note there is a need to put as many features into the system as possible.
[posted by Jill Rouleau on chromis blog] It is well documented how to configure hardware independent faxing using HylaFAX and IAXModem with Asterisk and trixbox, but not for Switchvox. So I created a Switchvox-specific walkthrough. This configuration was tested on a Shuttle PC, then implemented on a Xen virtual machine. It should work with any Linux distribution and hardware that support Hylafax. Source code is available at: http://sourceforge.net/projects/iaxmodem/ and http://www.hylafax.org/content/Download. If you compile from source, you will need to create a few directories manually, such as /etc/iaxmodem/ and /var/log/iaxmodem/ with files /var/log/iaxmodem/iaxmodem and /var/log/iaxmodem/ttyIAX0.
Visual Dialplan Professional 3 comes with support for Apstel Integration Server, powerful application server that extends Asterisk dialplan with new functionality and simplifies work with all major databases (MS SQL, MySQL, Postgres, Sybase, HSQL, etc.) and email servers (Gmail, Sendmail, Exim etc.).
Newly added building blocks provide an intuitive interface to easily execute SQL queries, send email messages and do more from the Asterisk dialplan.
Beside these components Visual Dialplan Professional 3 comes with full featured SQL query builder, email template designer, and several new components like Fax for Asterisk and other.
The Asterisk Development Team has announced the 2nd set of release candidates of Asterisk for versions 1.4.34 and 1.6.2.10. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
These release candidates address issues that were reported by the community and resolved since the last round of bug fix releases.
The following is are issues resolved in these release candidates:
* Fix problem with RFC 2833 DTMF not being accepted.
Watch this new video from Digium to learn how you can get started with Asterisk! During this Steve Sokol presentation you will learn:
What is Asterisk?
•What can I do with Asterisk?
•Which version should I use?
•How do I get started?
i6net.com please to announce a new important feature for voice & video services over our next release of VXI* 5.1: Flash/RTMP for Asterisk / VXI* is an add-on channel driver for Asterisk-based PBX systems. Adding Flash/RTMP for Asterisk to any Asterisk server enables complete access to manage bi-directional video calls from a Web browser with a Flash player.
Real Time Messaging Protocol (RTMP) is a proprietary protocol developed by Adobe Systems for streaming audio, video and data over the Internet, between a Flash player and a server. Adobe Flash, is the industry-leading web application environment, present in web browsers on 99% of the world’s computers. Flash can access the webcam and microphone on a PC and works through any firewall.
Kolmisoft announces release of MOR version 9 - Advanced Softswitch with Billing & routing functionality that is dedicated for companies who would like to start any kind of VoIP business and for small and mid-size telecoms who would like to switch to more advanced softswitch or billing platform.
Probably the biggest change in MOR 9 is Payment Gateway addon. While basic MOR license allows PayPal, Webmoney, Ouroboros and Cyberplat online payments, release of Payment Gateway addon brings completely new popular payment gateways like:
* 2Checkout.com
* Authorize.net
* Google Checkout
* Moneybookers
* PayPal PRO
Elastix was the first distribution to include an Open Source Call Center module with a predictive dialer. This module can be installed from the same web-based Elastix interface through a module loader.
Current version: 1.5-3.2
The call center module can handle incoming and outgoing campaigns. Some of its features are:
Technetworks has developed the first fully functional CSTA III XML interface for open source Asterisk, called the Asterisk Connect. The interface currently supports events and call control including single step and attended transfer functions.
Until today interfacing Asterisk with major business applications has been difficult, time consuming and in many cases the end result was far from stable under heavy load. Not anymore! With Asterisk Connect we present an industry standard CSTA III XML interface to interface with. Asterisk Connect functions as a gateway/proxy and translates the CSTA III protocol to Asterisk operations/events and vice versa.