Showing posts with label telephony. Show all posts
Showing posts with label telephony. Show all posts

Thursday, June 16, 2011

IP Telephony Implementation at Villa Maria National University

Villa Maria National University is an education organization established at Villa Maria city in Cordoba State, Argentina. Two years ago directives of the university order the transfer of the existing telephony network to an IP platform, with a preliminary evaluation of technical and economical feasibility.

Universidad Nacional de Villa María

Argentina

Mariano Martín

Villa Maria National University is an education organization established at Villa Maria city in Cordoba State, Argentina. Two years ago directives of the university order the transfer of the existing telephony network to an IP platform, with a preliminary evaluation of technical and economical feasibility.

Different versions of Asterisk were analized in several Linux distributions at virtual environments. At the end the solution chosen was an integral/complete distribution.

The initial tests were performed on Elastix v1.2 32 bits in a real server using Digium clone cards (modems with Motorola/Intel chipset) with FXO ports connected to the PSTN. At this stage Softphones and IP phones were use for users.

Once the tests concluded, the authorities of the University took the decision of change the platform in their policies.

A Network adjustment was made in order to work with VoIP, also a there was a replacement of some switches by others with Power over Ethernet.

The infrastructure of the main Communication server was installed in an Intel Xeon/ HD 500 GB server with Elastix v1.3 32 bits.

At first we install 10 extensions using Grandstream IP Phones model GXP2000.  We also used a Sangoma A200 card with 4 FXO ports to access the PSTN.  This implementation was executed at one of the extensions of the University Campus.

In the next stage we include a Sangoma A200 telephony card with echo cancelation to expand the telephony integration to a total of 24 FXO ports. With this expansion we create more users to the system that include 100 extensions at the Dean Headquarter using different IP phone models (GXP2000, GXP1200, GXP280, and BT200). We also added console operators for more than 100 BLF indicators.

In addition to the 100 users, 10 Grandstream GXV3000 IP VIDEO phone where included. For this purpose, we enabled video support and all necessary codecs.

In the following stages the University incorporated a new Elastix v1.3 de 32bits server with the same hardware characteristics. for the Central Campus The telephone integration was made with a Sangoma A200 card with 24 FXO ports.  Both servers were interconnected with IAX2 Trunks and a dedicated wireless link with 22Mbps and 100 extensions where added.

The main goal of having two servers with their own access to the PSTN is to avoid the lack of communication between the extensions if the link failed at some point.

For this functionality we create 15 users and used telephone adapter switches (ATA) for Fax machines.  After that we created 20 virtual FAX (Hylafax), all of them redirected to several emails. There is also an access to FAX from the PC with Winprint Hylafax.

Some of the following features where enabled:

A freepbx CUSTOM-CONTEXT module was installed to manage different access to the PSTN for internal groups.

Conference Call, Call transfer, Call Pick-up, Call forwarding, Ringing groups, IVR, etc…

A call billing feature was enable, this was possible by including a destiny code to the bill. A2Billing was configured in order to work with different users with a billing process and prepaid call system.

Currently there is a development in AGI-PHP to use a text to speech application to allow teachers and students to make consultations of a database. This prototype is already working.

This implementation is part of a project that includes a complete implementation of IP telephony and Unified communications at the Villa Maria University which include a lot of work and further conclusions.

Some load data where obtained by creating modifications in the PHP code of the following module to reports and graphics in Elastix:

/admin/common/graph_hourdetail.php

This made possible to obtain the MAX LOAD value of the server in concurrent calls.  We modified the code to obtain that value in a regular day and then for a complete month.  We got the following results:
Dean Headquarter IP-PBX Server

Dean Headquarter IP-PBX Server


Here you can appreciate that January is the month with the lowest activity due to the summer vacations and the higher activity was detected on December 2009 with 21 concurrent calls registered in the Principal server and 14 concurrent calls registered in the  Campus server.

Elastix allowed us to reduce the implementation times. It is also useful to the system administration allowing the addition of more users with an easy training thanks to the information available.

At the moment of the publication of this case we have 4 servers interconnected with IAX2 and SIP.

The total number of extensions is 250 and we are working on a main project that will include VoIP platforms of other Argentinean public Universities to our system through a Proxy SIP Server.  

The coordination of the telephony network that includes 18 Nacional Universities is performed at The Villa Maria National University and San Luis.

Monday, February 7, 2011

Asterisk SCF: Make Open Source Cloud Telephony Happen

It’s been two months since Asterisk SCF was announced in AstriCon 2010. It really brings a lot of expectation to the open source community and it’s worth our waiting. Asterisk SCF is still in heavy development and will not be expected to be beta-level software until some time in 2011. But are you ready when it hits the ground? If you would like to start developing on Asterisk SCF, please Visit the Asterisk SCF wiki  to find information on downloading and building the Asterisk SCF source, documentation on the architecture, and see the Roadmap.

asterisk-scfAsterisk SCF is a framework that allows developers to create real-time communications applications that include voice, video and text and that meet the demands of a full range of uses, from embedded applications to enterprise and carrier solutions. Asterisk SCF is architected to provide the highest levels of availability, scalability, extensibility, fault-tolerance and performance.

Asterisk SCF will be delivered as a system of distributed components that can be deployed in clusters on a single system or on many systems, transparently. The Asterisk SCF platform will support, as a part of its basic architecture, the full range of real-time IP communications, including video, multi-channel wideband and ultra-wideband audio, chat, desktop sharing and other media types that may arise in the future.

Asterisk SCF is not a replacement for Asterisk, the world’s most widely used open source voice communications platform. Digium and the Asterisk community are committed to the continued development and support of Asterisk, the telecommunications software.

asterisk · Asterisk SCF · AstriCon 2010 · digium · embedded application · open source

<< More Than Two Million Downloads of Asterisk in 2010

The First Open Source Telephony Conference in 2011 >>

Tuesday, December 21, 2010

The First Open Source Telephony Conference in 2011

The real beautiful thing about open source telephony is when everybody talks about it and benefits from it. I am happy to know that the VoIP88.com has announced the 1st open source telephony conference in Shenzhen China in the coming year. And I do think it’s also the 1st one in the worldwide community since it will be launched on 10th Jan, 2011.

voip88_logo

VoIP88.com is now the largest open source telephony technical forum in China and is authorized to release the Elastix Chinese Version to the community. They are now providing free technical support in its forum and Elastix solution nationwide in local language.

What to Expect in the Conference?
The first Open Source Telephony Conference will be focus on promoting the most popular open source telephony software in China including Asterisk, AsterCC, Elastix, freeSWITCH, FreePBX and so on. Meanwhile, you will also expect to meet the new solutions and products provided by the hardware manufacturers such as Digium, OpenVox, Sangoma.

Who to Meet in the Conference?
Well, you will meet VoIP providers, PBX developers, Call Centre Developers, System Integrators of Open Source Technology, Asterisk users, hardware providers.

Conference Speakers:
Doug Vilim from Sangoma Technologies Corp
Solo from AsterCC
Tony Zhang from VoIP88.com

Click HERE for the original conference news in Chinese.

AsterCC · asterisk · Asterisk China · China Asterisk Conference · digium · elastix · freepbx · freeswitch · open source · OpenVox · sangoma · telephony · VoIP · voip88

OpenVox Announces Free Support for Asterisk Configuration >>

Thursday, December 9, 2010

Gitex 2010:The Rise of Open Source Telephony in Middle East

You are expected to see a new face in Gitex 2010 Dubai this week: OpenVox has been presented in the Zabeel Hall in booth Z-P31. It’s the first time that OpenVox shows up in the Gitex in the Middle East Marketing.  And the some of OpenVox newly released products is being presented in this show including AE2410P, IPC100 for a better choice to build a best cost effective Open Source telephony solutions.


gitex-2010 GITEX TECHNOLOGY WEEK is at the heart of the global technology network. Experience a spectacular celebration of interactive features, a showcase of leading technology brands and an unrivalled range of networking opportunities as GITEX celebrates 30 years of industry integration and leadership. Review and select from thousands of leading technology brands and industry experts.


Who else is to see in Gitex 2010? DVCOM is presenting Digium, Snom,Patton in its booth. You are not limited with your choice of asterisk products since OpenVox has already been here for you.


Tuesday, December 7, 2010

Asterisk PBX Text-to-Speech

Text-to-Speech for Business Telephony with Asterisk PBX

It is expensive and wasteful to hire humans to answer or guide callers. This is when the beauty of automated natural sounding text-to-speech software is realized. Text-to-speech conversion with high-quality, natural sounding voices for hand-held, desktop, and server applications adds a lot of value to business telephony with Asterisk PBX.


Text-to-Speech voices are available in 6 languages and 18 different characters including "Allison" the standard Asterisk voice. These voices are sizable and can range from 4 MB for embedded applications, to 100 MB for server-based installations. Text-to-Speech voices are tested on all major telephony platform operating systems including: Windows Server, Windows XP, Linux (32 & 64 bit), Solaris (Sparc and x86), and Mac (PowerPC and Intel).

The benefit of using text-to-speech voices with hosted Asterisk solution is that you can select a voice that fits your requirement, optimize it for your application, or even have voices built for a particular character or delivery. The voices run on the text-to-speech technology are petite and swift enough to run on handheld devices or distribute over the network.
The text-to-speech voice facility is not only limited to the telephony industry. In fact it is used across sectors such as health, navigation, mobile, toys & games, Government, industry, assistive, and many more. Now with Asterisk VOIP PBX, text-to speech functionality is available for really affordable prices to small and medium sized businesses. Also, don?t forget to enjoy the benefits of the best application hosting solutions, such as hosted VOIP PBX, Exchange, SharePoint, etc., which has become the preferred way for companies to implement business software.


Monday, December 6, 2010

Text-to-Speech for Business Telephony with Asterisk PBX

Timesheet Management in Microsoft Project Server

If you have to have a disciplined working environment, then managing and maintaining a record of time spent on each tasks is important, isn?t it? Timesheets help record the start and end time of tasks, or just the duration. It may contain a detailed breakdown of tasks accomplished throughout the project or program. This information may be used for payroll, client billing, and increasingly for project costing, estimation, tracking and management. Nonetheless, this is where Microsoft Project Server and its Timesheet Management feature has a role. This feature in MS Project Server helps you to create and keep a tab on the timesheet of your project, thus making it easier for you to manage the timesheet of your work smoothly. It enables you to easily track the time you spend on a project's tasks, and the progress of those tasks towards completion. You can also enter time spent on administrative tasks, like vacation, travel, training, and sick leave.

Timesheet Management Features of Microsoft Project Server Hosting


? Create and Edit Timesheet
? Surrogate Timesheet (create on behalf of someone)
? Import tasks (with actual values) from project or create your own task entries
? Mark timesheet rows as billable for your invoicing needs
? Forward to Timesheet Manager or others for review
? Approve/Reject Timesheets.
? Merge timesheets received from team members to easily create a team timesheet
? Lock timesheet to block further edits
? Link to other objects like tasks, issues etc.


How to set up time reporting in Project Server?


? Set up time reporting periods: Time reporting periods are used on both the My Tasks and the My Timesheets pages, and must be set up before either page can be used.

? Configure timesheet line classifications: Timesheet line classifications are used to provide detail about project tasks.


? Configure timesheet settings and defaults: Timesheets can be configured in a number of ways that apply business rules to how users are allowed to track time. This is where you can turn on single entry mode.


? Configure administrative time categories: By default, Project Web App provides three administrative time categories: Sick, Vacation, and Administrative. Your organization may have specific categories that meet your needs.


? Configure task settings and display: Task progress updates can be configured in several ways to meet your organization's needs.


So get hold of this timesheet management feature of Microsoft Office Project Server to lower costs and increase the productivity and health of the company! What more, grab the best Application hosting solutions, such as Microsoft Exchange Server, SharePoint from Apps4Rent that too at cheaper rates!

Thursday, November 4, 2010

Infonetics Research reveals its predictions on VoIP telephony growth in 2010

Infonetics Research has announced some digits and estimates on expected 2010 VoIP telephony growth perspectives. Global VoIP service revenue is estimated to be $22.6 billion in the first half of 2010, compared as up eight percent over the second half of 2009, driven by stable growth in business and residential services throughout all regions.

“ The market for business Voice over IP services slowed during the financial turmoil of 2009, but we are now seeing a good uptick in new business across all segments, particularly in IP connectivity services," said Diane Myers, directing analyst for VoIP and IMS at Infonetics Research. An steadily growing amount of hosted VoIP providers are trying to integrate Unified Communication elements and mobility into their standard packages, thus fostering new levels of service differentiation as well. The total amount of residential VoIP subscribers expanded by 10 percent between the second half of 2009 and the first half of 2010. There are some estimates on U.S. VoIP revenue (consumer and business) at about $14 billion in total. To evaluate those numbers in strategic perspective, anyone should keep in mind that in 2007 and 2008, the last years, when Infonetics Research had comprehensive data, U.S. full scale communications revenues were in the range of $297 to $299 billions. It means that general VoIP revenue represents possibly two percent of total communications industry ones. That's dramatically less revenue than most of involved individuals probably would expect by this point. Notwithstanding the fact that residential VoIP services revenue growth is slowing due to pricing pressures throughout the globe, trends of growth continue to get presented in regions such as Africa and Southeast Asia that previously did not have a large residential VoIP provider base, as per Infonetics Research. In the first half of 2010, the IP connectivity segment of the business VoIP services market surged past the managed IP-PBX and hosted VoIP and Unified Communications services sectors to become the largest segment of the market. In other words, SIP trunking now is the single largest portion of the IP telephony service revenue stream.

Wednesday, September 15, 2010

Meet The 2600hz Project, The New Sound of Open Source Telephony

I really miss my Bluebox and the days then that would work.  Gotta love the $5 rRdbox from hallmark or the infamous Blotto Box that was no joke at all.  We salute you Captain Crunch!
 
Gigaom - Some of the core developers behind FreePBX — a well-known, open-source phone system — have teamed up and started The 2600hz Project, a commercial entity promoting a collection of open-source telephony applications and libraries. Today, they are releasing blue.box, a reworked version of open source FreePBX. The new venture is co-founded by Darren Schreiber and is also a subsidiary of newly formed VoIP Inc. The 2600hz Project received $250,000 in funding from an unnamed investor.

2600 Hz is the frequency that the phone companies used back in the day and was hacked by those seeking to make free long distance phone calls. In order to do so, one needed a device that generated the 2600 Hz tone, called the blue box. The new venture is an homage to that heritage.

From what I understand, the new company was formed after some disagreements between the FreePBX developers and the original backers of the project. FreePBX is a graphical user interface that sits on top of open-source telephony software such as Asterisk. FreePBX was promoted by Bandwidth.com.

Click Here to Continue Reading

Saturday, August 28, 2010

Meet The 2600hz Project, The New Sound of Open Source Telephony

I really miss my Bluebox and the days then that would work.  Gotta love the $5 rRdbox from hallmark or the infamous Blotto Box that was no joke at all.  We salute you Captain Crunch!
 
Gigaom - Some of the core developers behind FreePBX — a well-known, open-source phone system — have teamed up and started The 2600hz Project, a commercial entity promoting a collection of open-source telephony applications and libraries. Today, they are releasing blue.box, a reworked version of open source FreePBX. The new venture is co-founded by Darren Schreiber and is also a subsidiary of newly formed VoIP Inc. The 2600hz Project received $250,000 in funding from an unnamed investor.

2600 Hz is the frequency that the phone companies used back in the day and was hacked by those seeking to make free long distance phone calls. In order to do so, one needed a device that generated the 2600 Hz tone, called the blue box. The new venture is an homage to that heritage.

From what I understand, the new company was formed after some disagreements between the FreePBX developers and the original backers of the project. FreePBX is a graphical user interface that sits on top of open-source telephony software such as Asterisk. FreePBX was promoted by Bandwidth.com.

Click Here to Continue Reading

Sunday, December 13, 2009

Digium Asterisk World

Digium|Asterisk World
Submitted by Anonymous on Fri, 10/09/2009 - 14:18
Event Date:
Wednesday, January 20, 2010
Event Location:
IT Expo East - Miami
Event URL:
Digium|Asterisk World

Entering its third year, Digium|Asterisk World at ITEXPO is the conference that addresses "Everything Asterisk" for business users, resellers and executive decision-makers.

Come to Digium|Asterisk World to discover how Asterisk, the world’s most widely used open source telephony software, can save you money and empower you to create more flexible telephony solutions.

Asterisk, originally created and maintained by Digium, is the world’s leading open source telephony engine and tool kit. Offering flexibility unheard of in the world of proprietary communications, Asterisk empowers developers and integrators to create advanced communication solutions...for free. The first ever Asterisk event focused purely on Asterisk for business, the Digium|Asterisk World Conference & Expo is "Everything Asterisk" for the business user, reseller, and executive decision maker.

Digium|Asterisk World is focused on Asterisk business users and potential users who are interested in getting more in depth experience with Asterisk, and who also want to learn more from Asterisk adopters and see what's happening in the Asterisk market. Digium|Asterisk World at IT EXPO is a pure-play IP conference, where decision makers will learn they have a cost effective and powerful choice. Digium|Asterisk World gives these audiences the opportunity to experience the global reach of Asterisk and the world class solutions which can solve traditional problems with new technologies and techniques.

January 20-22, 2010
Collocated with ITEXPO East 2010
Miami Beach Convention Center, Miami, FL


Digium has led the Asterisk revolution in telecommunications. Hosting Digium|Asterisk World at ITEXPO East 2010 will allow us to share the vision and power of Asterisk with you – even if you are not completely familiar with open source.

Come get a taste of Asterisk, and learn how you can not only save money, but also how to use and create new technologies that never existed before.

-Mark Spencer, Digium

Saturday, December 12, 2009

Open-Source Telephony System

Asterisk is the world's most popular open source telephony project. Under development since 1999, Asterisk is free, open source software that turns an ordinary computer into a feature-rich voice communications server. Asterisk makes it simple to create and deploy a wide range of telephony applications and services.

Code for Asterisk, originally written by Mark Spencer of Digium, Inc., has been contributed from open source software engineers around the world. Currently boasting over two million users, Asterisk supports a wide range of telephony protocols. It includes rich support for the handling and transmission of voice over traditional telephony interfaces including analog lines, ISDN-BRI lines and digital T1/E1 trunks. Asterisk also features support for a wide range of VoIP protocols including SIP, IAX and H.323 among others. It supports U.S. and European standard signaling types used in business phone systems, allowing it to bridge between next-generation voice-data integrated networks and existing infrastructure.

Asterisk is released as open source under the GNU General Public License (GPL), and it is available for download free of charge. Asterisk® is the leading open source telephony project and the Asterisk community has been ranked as a key factor in the growth of VoIP.

Asterisk is like an erector set or a box of Legos for people who want to create communications applications. That's why we refer to it as a "tool-kit" or "development platform". Asterisk includes all the building blocks needed to create a PBX system, an IVR system or virtually any other kind of communications solution. The "blocks" in the kit include:

* Drivers for various VoIP protocols.
* Drivers for PSTN interface cards and devices.
* Routing and call handling for incoming calls.
* Outbound call generation and routing.
* Media management functions (record, play, generate tone, etc.).
* Call detail recording for accounting and billing.
* Transcoding (conversion from one media format to another).
* Protocol conversion (conversion from one protocol to another).
* Database integration for accessing information on relational databases.
* Web services integration for accessing data using standard internet protocols.
* LDAP integration for accessing corporate directory systems.
* Single and mult-party call bridging.
* Call recording and monitoring functions.
* Integrated "Dialplan" scripting language for call processing.
* External call management in any programming or scripting language through Asterisk Gateway Interface (AGI)
* Event notification and CTI integration via the Asterisk Manager Interface (AMI).
* Speech synthesis (aka "text-to-speech") in various languages and dialects using third party engines.
* Speech recognition in various languages using third party recognition engines.


This combination of components allows an integrator or developer to quickly create voice-enabled applications. The open nature of Asterisk means that there is no fixed limit on what it can be made to do. Asterisk integrators have built everything from very small IP PBX systems to massive carrier media servers.

Asterisk can be configured as the core of an IP or hybrid PBX, switching calls, managing routes, enabling features, and connecting callers with the outside world over IP, analog (POTS), and digital (T1/E1) connections.

Asterisk runs on a wide variety of operating systems including Linux, Mac OS X, OpenBSD, FreeBSD and Sun Solaris and provides all of the features you would expect from a PBX including many advanced features that are often associated with high end (and high cost) proprietary PBXs. Asterisk's architecture is designed for maximum flexibility and supports Voice over IP in many protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.

It can also be built out as the heart of a media gateway, bridging the legacy PSTN to the expanding world of IP telephony. Asterisk’s modular architecture allows it to convert between a wide range of communications protocols and media codecs. Asterisk as a feature/media server.

Need an IVR? Asterisk’s got you covered. How about a conference bridge? Yep. It’s in there. What about an automated attendant? Asterisk does that too. How about a replacement for your aging legacy voicemail system? Can do. Unified messaging? No problem. Need a telephony interface for your web site? Ok.