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. @NerdUno 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 http://t.co/akyNqFj
PIAF RED download sites;

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

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

OpenSIPS

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:

• Arno's IPTABLES Firewall
• Traffic Shaping
• IPSec, OpenVPN and PPTP VPN Support
• Caching DNS Support (dnsmasq)

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