SAN FRANCISO, CA — (Dec. 2010) – As part of its ongoing commitment to the FreeSWITCH community, VoIPKB is pleased to announce Official FreeSWITCH Trainings have gone worldwide. The first of its kind, this is an in depth course on FreeSWITCH and its related modules.

FreeSWITCH Training Bootcamp is a three-day, hands on course offered internationally. Each student will utilize an independent FreeSWITCH workstation/server, complete with high definition phones for testing, and will experience setting up and manipulating FreeSWITCH for use in PBX and carrier-grade services. Darren Schreiber, co-author of the FreeSWITCH book, leads the classes in San Francisco, Dallas and Australia.

Additional classes are planned for France, Brazil, Germany, Saudi Arabia and China. During the class, students will learn, step by step, how to configure, manage and program the FreeSWITCH telephony softswitch in great detail. This FreeSWITCH Bootcamp accelerates learning for students with a Linux background who want to participate in the explosive growth of open-source voice-over-IP (VoIP) telephony.

The next upcoming bootcamp will be held Jan 17th – 19th, 2011 in Sydney, Australia. More information and registration is available at http://voipkb.com/.

Topics that will be covered in this course:
1. Install FreeSWITCH on a Linux Machine
2. Understanding configuration files and the default configuration
3. Call authentication and routing basics
4. Integration modules
5. Understanding presence
6.Load balancing and high availability
7. FreeSWITCH Internals
8. How to debug and troubleshoot FreeSWITCH
9. Building Custom C Modules
10. Advanced Modules

Prerequisites:
The introductory course assumes that students are already familiar with Linux, Linux administration, networking and network administration including TCP/IP and UDP/IP. Specifics of Linux and networking as they apply to FreeSWITCH installation and configuration are covered.

About VoIPKB VoIPKB is a VoIP knowledge base and professional services organization located in San Francisco, CA. Specializing in open-source telephony, the organization includes some of the sharpest talent in the open-source VoIP space. The company focuses primarily on soft switching technologies for carrier-based services, ISPs, SaaS application services and managed service providers.

About FreeSWITCH
FreeSWITCH is an open source software communications platform designed to facilitate the creation of voice and chat driven products scaling from a soft-phone up to a soft-switch. It is released under the MPL free software license and can be used as a simple switching engine, a PBX, a media gateway or a media server to host IVR applications using simple scripts or XML to control the call flow. Its core library, libfreeswitch, is capable of being embedded into other projects, as well as being used as a stand-alone application.

Open source developers from the East and West, North and South on planet Earth will converge on Chicago, Illinois in August 2011 for Cluecon. Make plans to be there.

Visit http://www.techistan.com/category/channels/asterisk-channels/ for more training in Urdu.

Visit http://www.techistan.com/category/channels/asterisk-channels/ for more training in Urdu.

Visit http://www.techistan.com/category/channels/asterisk-channels/ for more training in Urdu.

Visit http://www.techistan.com/category/channels/asterisk-channels/ for more training in Urdu.

Visit http://www.techistan.com/category/channels/asterisk-channels/ for more training in Urdu.

Visit http://www.techistan.com/category/channels/asterisk-channels/ for more training in Urdu.

Visit http://www.techistan.com/category/channels/asterisk-channels/ for more training in Urdu.

Hello and Hajimemashite.

DIDx (http://didx.net) offers wholesale inbound telephone service using the VoIP standard SIP (session initiation protocol). The service provides users with local PSTN numbers. And you can receive calls over the Internet from up to 60 countries. This document will explain how to utilize Brekeke PBX with DIDx.

Setting Up A DIDx Account

Please obtain an account from DIDx Website http://didx.net. And point the DID number you got to your PBX global IP address,like, DID@pbxIP, or pbxExt@pbxIP, then the call to the DID number will be routed to your pbx.

Brekeke SIP Server Dial Plan Settings

Dial Plan can accommodate the unique requirements of each VoIP service provider.
The following dialplan may need when auth is set on at Brekeke PBX bundled SIP server.

Receiving Calls from DIDX Number

This Dial Plan needs to be placed before “To PBX” (default dial plan)

Brekeke PBX ARS Settings

The following ARS rule may be needed to route calls from DIDx to a Brekeke PBX extension

One could argue that the greatest impact of (snom’s dedication to) interoperability and open standards can be measured in dollars or deployments or even ease of use. But in the case of EsSalud, the national healthcare network of Peru, the greatest outgrowth of interoperability and open standards is scalability.

In 2007, as the national Government of Peru began a substantial modernization and expansion of its hospital and healthcare system, technology officials decided to move from a traditional TDM phone system to an all Internet protocol (IP) network capable of handling voice, data and video across the country. With the help of SUMTEC, the largest technology and communications systems integrator in the country, EsSalud embarked on a mission to deploy an Asterisk-based IP voice communications system across hundreds of locations. After testing several endpoint solutions, SUMTEC and EsSalud chose a single desktop phone vendor – snom.

Today, SUMTEC and its installation partners have installed 10,000 snom phones across the 400 locations, with 90 percent being snom 300 desktop phones and the remaining snom 370s. EsSalud also took advantage of snom’s unique SIP-based paging unit, the PA-1, which is installed in several locations as a public address system.

SUMTEC and EsSalud will continue to roll out snom phones and new systems to the remaining 30 percent of locations that have not been upgraded, as well as include the system in any new construction.

With a project of such magnitude, scalability was, and continues to be, the catchword of the deployment, but that doesn’t discount costs. After testing phones from GrandStream and Polycom, SUMTEC and EsSalud found snom phones provided greater interoperability at a lower price point. Officials report the choice of snom phones has saved EsSalud more than $1 million (US) to date.

While the project has been heralded a success for EsSalud, SUMTEC, snom and other technology participants, it has also had a major impact on the modernization of Peru itself. The expansion and modernization of the hospital and healthcare network was a major issue in the presidential elections of 2006, and the successful execution of the initiative has helped President Alan Garcia Perez fulfill his campaign promises. In fact, Garcia has personally inaugurated each new hospital and in one instance placed a test call to Jorge Lay Castro, Principal of SUMTEC using a snom 300 desktop phone.

In the end, EsSalud has been able to provide modern IP telephony and all its features to hundreds of locations using thousands of snom endpoints and bolstering Peru’s national infrastructure at the same time.

Some of the immediate benefits include:

- Easy provisioning of new lines, from central site or across branch locations;
- HD audio quality across the entire snom product line;
- Interoperable fail over from OCS to other environments in case of network disruption;
- Interoperability across many different SIP products.