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Integration of Android Application with ICTBroadcast

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Android application for ICTBroadcast allows to create voice recordings, contact groups and also create and run voice campaigns. A screenshot of the main screen is shown below. User of the ICTBroadcast should have a username and password to be identified as authenticated user. A URL to REST web services of ICTBroadcast is also required. For example, http://myictbroadcast.com/rest

Below are different features of the application and how they work. These components include settings, campaigns, recordings and contacts.

Settings

First things after running the application is to configure it. Settings tab allows user to set their username (Account Name), password (Secret Key) and set the location of the ICTBroadcast server (REST API URL). This is one time step usually when using the application for the first time.

Recordings

Recording allows to create new voice recordings and upload them to the ICTBroadcast server. User can also listen its created voice recording before uploading. Once user is sure that his/her recording is fine, then he/she can choose to upload that recording. Uploading requires to enter the name of the recording to be identified.

Contacts

Contacts allow upload all the contacts the android phone to as new contact group at ICTBroadcast.

Campaigns

Campaign form allows to create a voice message campaign in 3 simple steps. Set a name for the campaign, select voice recording and select contact group from the dropdown and finally clicking Run campaign button to run campaign. At the moment, there is no option to monitor statistics and summary of the campaign in android application.


ICTBroadcast, Using REST based API to automate your business

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ICTBroadcast recognize that the modern SME utilizes a wide spectrum of software and web-based applications to facilitate the needs of the business. We also understand the frustration caused by the lack of compatibility between some of these applications and the decrease in efficiency or increase in cost that inflexible software system can cause.

For this reason ICTBroadcast uses REST based API to ensure communication and automation between our products and third party applications is a seamless and simple process. Customers can make the most of third party applications without compromising the quality of service ICTBroadcast provides.

There are four type of API class available to ICTBroadcast. With these the user can define all the functions, which need to be drawn from outside the ICTBroadcast system. The classes are as follows:

API_User
API_Contact
API_Campaign
API_Recording

The functions within each class allow complete integration of the system and can be used to rapidly establish tools to add or delete users, make payments to user accounts, commence or conclude a campaign and much more.

Because of the seamless integration it is a straightforward task to manage the creation of users across multiple applications without the need to manually duplicate tasks. In this case the programmer can make use of the API named “user_create”.

A full list of the API’s available can be found here (link to http://www.ictbroadcast.com/news/using-rest-api-integerate-ictbroadcast-...). Other frequently used API’s include addition of credit (or the making of a payment to a customer account), the return of campaign status information and creation of a new campaign for a particular customer.

To set up ICTBroadcast for use with a third party application is a short task that only requires to write client-side code to invoke these REST APIs in third party applications to communicate with ICTBroadcast. Once the client-side code is working, the automation may be undertaken.

Further benefits of the REST based API system are seen in the platform that stems from such a decoupling. This means that significant scalability becomes possible and a vast range of desktop, web based, and smartphone applications (such as Android Client of ICTBroadcast) become available for full integration into the ICTBroadcast system. The expansion of the system in this manner places ICTBroadcast as a highly competitive and efficient way to operate. This is in distinct contrast to the use of native libraries that might otherwise be used in the absence of a REST based API.

In summary, the scalable, low coupling and custom-design rich features reduce overheads by placing fewer demands on your programming team. ICTBroadcast can harness the power of third party applications and the full range of features can be automated without a significant effort. The savings in time and money allow your enterprise to focus on value driving activities rather than grappling with uncooperative infrastructure. Contact ICTBroadcast today to learn more about our solutions to your business needs.

ICTBroadcast, Creating dynamic messages with auto dialer and custom tokens

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One of the many benefits of running the ICTBroadcast system for an SME is the flexibility and degree of personalization that the system allows. Dynamic messages can be programmed to run via the auto dialer during a campaign tailored to your existing client database or as part of standard operating procedure. The creation of either an SMS or email template is a straightforward and simple matter. This can be quickly accomplished and refined further at anytime during the campaign, or in the course of normal business operations.

Predefined tokens, or variables, for client details such as the clients name, address, phone number or any other custom token are inserted into your SMS or email template within the ICTBroadcast editor. Once the template is defined the auto dialer will deliver a tailored message to each of the recipients in the database or generate a specific message in response to a user action.

This system is simple to manage and can be used in numerous situations. For example, the system can be used to generate a confirmation message for successful subscriptions, as part of a marketing campaign or to acknowledge receipt of data from your customers. Custom tokens function by creating a tailored message specific to your client or customer.

An example of custom tokens in use can be seen in the following message:

“Hi, [contact:first_name] [contact:last_name] this is to confirm that Company Name have received your message and one of our specialist [contact:custom1] customer service representatives will be soon be in contact”

In this example there are three custom tokens:

[contact:first_name]
[contact:last_name]
[contact:custom1]

These tokens, and all the other tokens available for use, are populated with reference to data submitted by your customers. In this example the data will correspond to the contact information a customer has provided. This data can be uploaded via a web interface or alternatively the customer is able to make use of enabled REST API, this allows integration of ICTBroadcast with your database. Database integration is a simple proceed and allows specific customer information to be dynamically retrieved so the customization process becomes automated.

There are three key benefits of using such a system.

Primarily, the automation process can save a significant amount of time what would otherwise require dedicated monitoring and response. In this manner the process of building a relationship with your client base is automated.

Secondly, the system serves a method to strengthen relationships with your client base. By acknowledging communication from the client in a tailored manner not only does client confidence in your company increase but also there are more opportunities to provide information about your products or services.

Thirdly, the use of customized interaction with clients can lead to greater conversion rates. By offering a tailored service that is personalized to each visitor to your site the rate of conversion to a paying client or customer significantly increases. Furthermore, the retention of existing clients or customers can benefit.

The easy use of custom tokens is just one of many reasons why ICTBroadcast is the integrated system of choice for our rapidly expanding client base.

ICTBroadcast, the unified auto dialer compliant with Telephone Consumer Protection Act of 1991

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The ICTBroadcast multi-tenant unified communications package is a powerful telemarketing suite that can dramatically enhance an SME’s capacity to boost efficiency or promote a product. However, the misuse of the product - or any other telemarketing solution - can cause a business to fall foul of consumer and data protection laws and regulations. Therefore, it is vital to become familiar with the local regulations to avoid substantial penalties or heavy fines.

The Telephone Consumer Protection Act (TCPA) of 1991 is most commonly referenced federal statute that telemarketers need to be aware of. TCPA includes rules such as which locations may be called and when those calls can be made. The provisions of TCPA allow for a penalty of $1,500 per violation of any of the numerous stipulations. With an auto dialer system having access to vast database of contact information the potential for significant financial damage is easily foreseeable. ICTBroadcast is fully complaint with TCPA 1991 and allows telemarketing campaigns to achieve their aims within the requirements of the law and without fear of accidental breach of the extensive regulations.

The precise requirements of TCPA and Federal Communications Commission (FCC) state four distinct requirements that must be met for compliance. The exact wording is as follows:
(1) prior express written consent for all autodialed or prerecorded telemarketing calls to wireless numbers and residential lines; 
(2) allow[s] consumers to opt out of future robocalls during a robocall; 
(3) limit[s] permissible abandoned calls on a per-calling campaign basis, in order to discourage intrusive calling campaigns; and 
(4)exempts prerecorded calls to residential lines made by health care-related entities governed by the Health Insurance Portability and Accountability Act of 1996.
ICTBroadcast incorporates these requirements into your system via integration with the Do Not Call (DNC) registry and user defined limits on campaign variables.

The DNC registry is maintained by the Federal Trade Commission and allows consumers to opt out of unsolicited telemarketing calls. Failure for an SME to honor the DNC registry and its requirements can have a catastrophic impact. Even large corporations can fall foul of the stringent rules. In 2014, the third largest U.S telecommunications company, Sprint Corporation, were ordered to pay $7.5 million as a result of their non-compliance with the DNC registry requirements. ICTBroadcast offers global and local integration with the DNC registry to prevent such costly errors. Furthermore, contacts can be given the option to opt out from future marketing calls and have their details removed from the list by setting up a straightforward key-press operation that further prevents non-compliance with regulations.

ICTBroadcast makes compliance with TCPA and other legislation designed to prevent the misuse of telemarketing products a simple task. The risks for failing to comply with local laws and regulations are substantial, and given the sizeable penalties possible the ongoing viability of an SME can be severely impacted. In addition to the previously mentioned compliance features ICTBroadcast allows the user to set a limit upon the number of times a customer will be called during a campaign. Persistent phone calls are a key factor in the decision of a customer to complain to regulating authorities. By reducing the nuisance factor ICTBroadcast can improve general public relations for your business.

In summary, use of the ICTBroadcast multitenant unified communications package facilitates compliance with TCPA and comparable legislation in a number of ways. Limits upon call frequency and timing can be set, automated database opt-out can be enabled and furthermore global and local DNC lists can be intergrated to ensure consumers who have expressed a preference do not receive calls.

This feature of the ICTBroadcast package is another of the many reasons more SMEs are selecting our software to facilitate their communication needs.

Open source and ICT working together for social justice

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The recent phenomenon of Information Communication Technology (ICT) – and in particular Open Source (OS) – as a tool to reach community goals such as social justice and public vigilance has reached new heights during the past couple of years. ICT4D, or Information Communication Technologies for Development refers to the use of ICT and OS in socioeconomic, international or human rights development projects. All major areas of development work can be enhanced by the appropriate deployment of ICT and a cursory Google search reveals the success of many of these ventures to date.

The OS movement was created in response to efforts by software manufactures. These manufacturers sought to take ownership of code created by teams or individuals. However, it was argued that code should freely available for use and for modification, companies seek to prevent this by exercising copyright privilege and limiting the rights a buyer has over their product. OS software is created by individuals or team who are often not part of the same organization and the final product is distributed freely. Furthermore non-technical experts can collaborate in the development of specific solutions. Participation in OS projects stems not from a desire to enrich oneself financially but from more community orientated aims such as peer recognition or the work itself as enriching on a personal basis. Participation in OS projects is supported by academic theory – such as Maslow’s Hierarchy of Needs – that shows that people are motivated by factors other than money. Reasons cited for participation in an OS project by programmers are often ones of intellectual fulfillment or peer recognition. A central release schedule is maintained for some OS products. This ensures users are aware of potential bugs and fixes. Also these central registries provide a change history of the code and allow programmer to tackle areas that need improvement. Users of OS programs are of course free to modify the code to suit their purposes and do not have to pay the license fee or royalties to traditional developers.

The primary concerns of current development projects such as water access; education, access to information and the empowerment of civil society have all received a much-needed boost from support by ICT developers and the use of OS software. Ready access to mobile technology and the lower costs involved - when compared to 10 years ago - have made mass participation more feasible and allows projects to reach a critical mass at a much more rapid rate than was historically possible. ICT and OS are a strong fit for the purposes of ICT4D because the large support infrastructure and expenses necessary to maintain the network are not present.

ICT has become a permanent feature in monitoring the exploits of repressive regimes and, for drawing public attention and support to counter campaigns. The ubiquitous nature of mobile technology allow it to t become the most powerful weapon against governments who have ceased to act in the interests of their citizens. Speaking to the British newspaper, The Guardian, a spokesperson for the Non Governmental (NGO) Tactical Technology Collective said, “While computers and the Internet can be extremely powerful tools for advocacy, they also present new risks for a group that is already, in many cases, quite vulnerable. As more advocates have begun to rely on digital technology to achieve their outreach, data-collection, information design, communication and mobilization objectives, these risks have become more severe."

OS components of ICT are a seemingly natural fit with this trend. The transparent nature of OS chimes well with a desire by many of these advocacy organizations to cast light onto the darker regions of government business. Examples of ICT projects in action include Daraja, an NGO operating in rural Tanzania. Their “Raising the Water Pressure” project encouraged people to use SMS to report faults with water points and to put pressure on the government to respond in more rapid manners to problems of this nature.

Other examples of ICT4D include the use of the technology in areas recently damaged by natural disasters. When an earthquake struck Haiti in 2010, much the communication infrastructure was destroyed or severely damaged. Several ICT solutions were used to aid in relief efforts including crisis mapping to identify and prioritize areas of need, the use of OS tools to monitor inventory levels and co-ordinate the relief effort and restoring Internet connectivity to local NGO’s by Inveneo, a non-profit organization who deliver ICT solutions to the developing world. ICT4D provided creative solutions at a low-cost to rapidly ease the suffering of those effected by the disaster.

OS and ICT can be utilized together to merge communication and information technology. This combine results in flexible solutions and the ability to customize OS programs in a manner specific to the ICT4D project. For example, data validation can serve a vital role in monitoring local elections, especially when the potential for electoral fraud is high. Monitoring projects can be established to identify unusual trends and to make the government aware that they are under scrutiny. Recent elections in Kenya, Nigeria and Georgia have used ICT4D projects to ensure adequate scrutiny. Alternatively app’s can be developed to make qualitative data more qualitative. However, caution must be exercised in these projects. Exposure of activists is a significant risk and even encryption may not always provide sufficient protection. The creation of more anonymous reporting systems is one possible route around this concern. Already one system operational in India allows users to send an SMS to a central number, which then passes on the message in an anonymous manner. Projects like this encourage the community to share solutions and then implement them via OS resources. Common objectives such as social justice are thus brought into reality.

ICT Innovations working to release ICTFax Version 3.0

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ICTFax Version 2.0 was released based on Plivo communications framework but later on Plivo has postponed supporting their project causing serious issues to community members those were using plivo in their systems as well as ICTFax users.

ICTFax users faced many issues those were related to plivo communications framework but there was no support. We, at ICT Innovations, after observing all this started working to replace plivo with our own custom framework so that ICTFax users will not face problems in future and now time has come to announce that we have completed most of the work and we are testing new framework with ICTFax and will release it soon.

ICTFax Version 3.0 will be released replacing existing plivo framework with our own ICTCore that will make ICTFax more flexible, stable and powerful solution. ICTFax previously relied on plivo to connect with FreeSWITCH. But now, ICTFax uses built-in ICTCore that connects directly with FreeSWITCH. ICTCore, apart from other features, also contain applications for sending fax and receiving fax functionality. In near future, it will be extended to contain more applications like play audio, call record, text to speech and many others.

New version also removes any need for the contributed drupal modules like Mailhandler and Feeds. ICTCore handles all the hassle itself now. No extra-ordinary configurations are required now. This means that configuring email to fax and fax to email functionality becomes much more easy.

Top 5 VoIP Mobile Apps

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VoIP mobile apps are in vogue and there are specific apps for different types of VoIP services. The smartphone market holds great potential for the VoIP technology and it is with the mobile apps that this market can be tapped. There are several VoIP apps that have become popular among the mobile phone users. In this article, we are going to look at some of the most popular apps that hold a lot of promise for its users:

Viber

One of the best applications for android smart phone is to use VoIP. This app routes the calls made by the user through data network instead of the service provider. Viber is considered among the best ones and the popularity of this app is just touching sky. At first it was regarded as the replacement of texting services, but now Viber allows its user to place a voice call anywhere in the world for free using data or Wi-Fi network; although the person to whom the call is placed must also be a Viber user. If you want to place calls to a landline phone using VoIP, then it also can be done by using Viber Out feature but these calls are charged with a minimal rate and vary according to the country you are calling. Viber is available for Android, iOS, Blackberry, Windows Phone etc.

Google Hangouts

The next one in the list the app called Google Hangouts. This is an interesting app in development and this app offers free voice calls to some its users and some are charged for the voice calls. The users, who are using iOS version of the app, can place voice or video calls to any user of Hangouts but who are using the Android version of the app can only place video calls for free and the voice calls will be diverted back to the carrier and they will be charged with normal plan. This app is available for iOS and Android platform. Using Hangouts, you can call from your computer and if you are calling directly to a landline, it is free as long it is domestic and the international calls are chargeable and affordable.

Skype

It is among the first ones in the market. This app is much discussed than the others. It offers its users to place free video and voice calls all over the world. It is among the most reliable apps to use VoIP for calling. This application is completely free and it is available for desktop OSes along with other platforms like android, iOS and windows phone. The one feature that is totally unique about this is that it supports large group video calls which other applications of this category do not offer. It also offers free texting services.

MagicApp

MagicJack is the app that makes a home phone transported to a phone that can use VoIP. But this MagicJack also has launched a mobile application with the name MagicApp. This app is the apps in the market which routes voice calls over data or Wi-Fi network and it also offers very low rates for calls for the citizens of US and Canada. But the calls made to other users of the same app are completely free. The app offers a special number to its users.

Vonage Mobile

The Vonage Mobile app is also one of the most popular apps in the market to place calls routed out of the carriers’ barrier and connect through Wi-Fi or data network. But the main feature is that this app allows its users to retain and use their own numbers. Voice, video calls and texting to other Vonage users is completely free while domestic calls to US numbers are also free but international calls are minimally charged.

About The Author

Michelle Patterson is an avid technology blogger and writes extensively about IP/VoIP and Unified Communication. She works with some leading companies to understand the trends of these modern communication technologies.

Home Service Providers

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With the introduction of electronic switching in 1960, telephone slowly went digital. The development of digital data communication such as internet, allowed digitizing data and transmitting it in real-time across networks. This system follows certain set of protocols like Internet Protocol (IP). For telephone, it is voice over internet protocol (VoIP). VoIP replaced almost all telephone networks. IP telephone uses high-bandwidth connection and special transmission equipment via internet.

VoIP

Voice over internet protocol is implemented in various ways through proprietary and standard protocols. H.323, MGCP, SIP, RTP, RTCP etc. are some of the protocols used today. H323 is one of the first used protocols that made its way in LAN services. However, new protocols like MGCP and SIP have replaced it. Today, Session Initiation Protocol SIP has wide spread implementation.
A VoIP phone is necessary to connect to this service. It is done in following ways:
1. VoIP phones connect to IP networks using Ethernet or Wi-Fi. This is the traditional way of connecting.
2. Sometimes they use analog telephone adapter. It connects network to conventional analog phone systems.
3. It uses electronics and firmware through a modular phone jack.
4. Softphone is most modern version, which is installed on a computer that has microphone and speaker set.
5. Software has tradition dial pad on the display, which can be controlled by simple keyboard and mouse.

Service Providers

Almost all telecommunication service providers have now switched to VoIP and they connect old telephone network to public IP networks. This is called IP backhaul. Today, with increasing use of smartphone VoIP is handy. All smartphone and Wi-Fi supporting devices have SIP built in its firmware.
There companies like Lingo, Ooma, Vonage etc. who are popular for providing VoIP services for home based customers. These companies provide services like voice, fax, SMS etc. through public internet. Early VoIP service providers offered business and technical solutions, which used similar telephone network technology. Second generation of service providers started providing services to private users like Skype. Sometimes they provide free call services and charge while accessing to other communication networks. Third generation of provider like GTalk has use federated VoIP.

Advantages of these services

Voice over Internet Protocols has large number of benefits over traditional PSTN.
Due to bandwidth efficiency, cost is lower. Business has seen 80 percent reduction in their expenses in phone communications.
VoIP solutions follow a business model, which provide unified communication services like calls, voice mails, emails, and video conference over web, and many more. There are two type of companies one who provider to large enterprise and other who provide service to small and medium range business phone services.
It supports both voice and data communication using same network, which reduces the infrastructure costs.
Unlike PBX and key systems, VoIP switches run on cost effective systems like computers.
With VoIP, customers get good interface system for their communications.

Quality of these services

Today IP networks are the most reliable services for communication than circuit switched networks, which was prone to mechanical errors. However, with IP services there no loss of data over network. It uses Quality of Service (QoS) guarantees to protect latency of data packets. Network router handles traffic over these networks. There is a chance of latency if traffic exceeds threshold for VoIP. But it is minimized by marking voice packets with method called DiffServ. VoIP usually wait for completion of transmission of previous packet before new is initiated.
Security for VoIP services is similar to those with the internet-based devices. VoIP uses firewalls for routing traffic, which is often challenge for a hacker. They also use network address translators to interconnect networks or the internet.

About The Author

Michelle Patterson is an avid technology blogger and writes extensively about IP/VoIP and Unified Communication. She works with some leading companies to understand the trends of these modern communication technologies.


How to become Internet Telephony Service Provider (ITSP) by using ICTBroadcast SP edition

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ICTBroadcast service provider (SP) edition has all the features of enterprise-grade software application for its customers who want to become Internet Telephony Service Providers (ITSPs) and want to provide their own broadcasting services (such as voice, sms, fax and email) to their customers. ICTBroadcast is productive, scalable, fault-tolerant and secure software and also comes with monitoring, integration, upgrading and support services. ICTBroadcast is unique communications auto dialer software that provides multi-tenancy, multi-service billing, CDR reporting, payment management, REST APIs and IVR designer features that makes it number one choice for the service providers who want to start their own broadcasting and telemarketing services.

Multi-tenant

For any online service provider system, an important feature is multi-tenancy. Multi-tenancy provides separate web portals for users and administrators without interfering with one another. It allows users to register at an online service for free or by paying subscription fee mostly using credit card. Once registered, user is allowed to use services until the subscription expires or balance decreases to minimum allowed to use service. It facilitates complete campaign management for every user. Each user can manage, monitor, and run multiple campaigns separately and independently.

ICTBroadcast has two built-in roles: user and administrator.
User: This is a user with limited privileges that can manage his contacts, manage voice recordings, manage fax documents, make payments, view his/her balance and view call detail reports. This role can also broadcast his message by creating different types of campaigns and interact with its customers/clients through voice calls, SMS messages, fax documents and emails. This user can also view campaign results and statistics.
Administrator: This role manages overall resources of system like managing users, configuring trunks, making payments on behalf of the users, adding new destinations, setting routes and most importantly allocating any number of extensions, DIDs and channels/lines to users. Administrators can monitor every user, its running campaigns, number of concurrent calls (channels) used and running balance of each user. In the latest version of ICTBroadcast, administrators now can also monitor system resources and servers statistics through graphs.

Multi-service Billing

ICTBroadcast has an integrated prepaid billing system and allows real-time control of calls and campaigns according to available funds. REST-based API takes billing and payments to next level by binding ICTBroadcast to a third party web portal. Multi-service billing offers customers a rich and online billing interface that provides access to consolidated accounts and electronic billing information. This is a powerful feature and its reporting capabilities enable users to take control of their communications services. Multi-service billing enhances the loyalty of customers, reduces operational and support costs.

Scalable and Fault-tolerant

ICTBroadcast handles all the complexities of a large scalable system in the back-end and hides these complexities from the users and administrators. As a user of the system, administrators and users don't have to worry about anything in the back-end. You just have to create your campaign and start dialing your contact group. ICTBroadcast can be scaled up to blast thousands of simultaneous calls. This is achieved through multiple powerful server computers working closely together to provide scalability and load balancing. Usually a single node can handle 500 simultaneous calls efficiently. Using this distributed approach, there is virtually no limit on scalabilty. One can increase and decrease server nodes to scale up and scale down respectively immediately depending upon ones needs.

Integration through REST APIs

Another key feature that any enterprise grade broadcasting software should have is application programming interface (API). ICTBroadcast provides REST based APIs for third party software integration. ICTBroadcast uses REST based APIs to ensure that automation between our product and third party application is a seamless. Customers can make the most of third party applications without compromising the quality of service ICTBroadcast provides.

Campaign Summary and Detail Reports

ICTBroadcast has chart based reporting that makes viewing and analyzing a campaign much easier. Campaign summary report gives an overall view of the campaign with statistics and graph. Campaign statistics include percentages of answered, failed and busy calls and other useful information about Do Not Call (DNC) and answering machine detection (AMD). Call Detail Report (CDR) gives complete billing related details about every call made during your campaign, how long it lasted, and how much it cost. CDR lists are the successful calls made where as campaign summary lists numbers and percentages of calls answered, failed and busy.

Becoming an Internet telephony service provider via the use of asterisk based business solutions: a layman’s guide

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An Internet telephony service provider (ITSP) provides Voice over Internet Protocol (VoIP) and digital telecommunications services to either residential or commercial enterprises. Digital telecom services eliminate the costs associated with traditional phone services and allow individuals or companies to make substantial cost savings with no loss of quality. With a general knowledge of ICT it is a relatively straightforward manner to set-up in business as an ITSP. At each stage of the process other services exist to enable the correct management of your telecommunications business. A common benefit citied by ITSPs for selection of this business model is the scalable nature of the service and how it allows an enterprise to build their business free from the traditional high capital expenditure costs of ownership that other telephony solutions would entail.

Stages to Consider when setting up as an ITSP

There are three stages to consider when setting up as an ITSP. These are the choices over which VoIP platform or switch to use, which VoIP carrier to chose and the selection of end user equipment. These decisions are all explained in more detail below. Furthermore, information is given on a number of suppliers at each stage to enable the potential ITSP to make and informed decision.

1. What does VoIP Platform or Switch do?

Firstly, you will require a VoIP platform or switch. This platform provides the class five features for your service. Class five includes features such as voicemail, hunt groups and call-forwarding. A number of companies who offer VoIP platforms are reviewed below to assist in the selection of this first stage in becoming an Internet telephony service provider.

2. Why is VoIP Carrier important?

The next requirement to establish as an ITSP is to engage a VoIP carrier. This is known as a class four service. Considerations to make when selecting a VoIP carrier are coverage and origination. For this reason it is important to check the footprint of a particular carrier in your target market before engaging their services. Factors such as quality and the availability of support should also be given careful thought before any decision is taken. Many companies will offer automated support as part of a free package and some companies even offer free support from human operators. With the large number of VoIP carriers in the market it is worth spending sometime comparing the features and support packages that come as standard.

3. Selecting End User Equipment

The final requirement is the choice of end user equipment or software. Some ITSPs may prefer to allow the client to use their own equipment whereas others provide this directly. Using an analog telephone adapter (ATA) allows existing equipment to be used in the ITSP system within substantial additional costs. Alternatively, an IP phone is a popular choice that gives access to the class five features directly from the phone unit.

Companies who can help you become an ITSP

Below, the details a number of companies who provide the services required to become an ITSP are given alongside of the factors that might affect your decision to engage these services.

Enswitch from Integrics

Enswitch from Integrics offer a soft-switch that provides commercial carrier grade telephony services. The features include multi-tenant hosted PBX, toll-free and number translation services. Using a service such as Enswitch allows an ITSP to scale their services and facilitate high volumes of concurrent calls.

A2Billing from Star2Billing

A2Billing is another service that is available for free and uses open source software for Asterisk to provide billing services for class four and class five softswitch services. The enterprise offers residential, commercial and prepaid billing. Getting started with a service like this is as simple as downloading the software and reviewing the enclosed documentation.

Multi-tenant PBX from Thirdlane

Thirdlane offer supported solutions for their multi-tenant PBX. The company has become a popular choice for ITSP because of their powerful end user features which include features that go beyond the basic VoIP package. The company also sells telephony equipment and other devices suitable for the ITSP to offer to their clients. Their built in support function aims to reduce the amount of reliance upon operators. Furthermore, the company offers pre-programmed support in multiple languages to allow an ITSP to expand their business on a global basis.

ICTBroadcast Service Provider Edition from ICTBroadcast

Integrated companies such as ICTBroadcast provide packages of services that include all the features an ITSP requires. ICTBroadcast are an established company who are known for the quality of their coverage and the level of support offered. ICTBroadcast even offers free technical support. Furthermore, they support Voice, SMS, Email and Fax broadcasting and deliver a wide range of real time metrics to enable ITSP to offer exception service to their clients. Their scalable auto dialer allows both billing and routing. It makes use of multiple nodes of open source Asterisk PBX that eliminate the bottlenecks a single Asterisk PBX can cause. A variety of packages are available that run across a multiple number of servers, further details of the specific software running on each server can be seen here. The open source architecture of ICTBroadcast allows ITSP enterprises to deploy solutions that are tailored to their particular market and to meet the needs of business customers who frequently require specific customization within their ICT systems.

QueueMetrics from Loway

For an ITSP data and user metrics are very important. The same is true for your clients. Those managing call centres use a range of metrics to assess the efficiency of campaigns and to ensure their employees are operating according to company protocol. QueueMetrics from Loway offer a range of services designed for the requirements of operations managers call centre team leaders, call agents and IT managers. The same software allows these distinct groups to see the metrics that are important for their role. For example, agents can see the inbound call traffic volume and data on the calls they are handling. Team leaders are given access to data on agent availability; real time reporting and remote monitoring for call quality purposes.

Summary

In summary, becoming an ITSP and generating a sustainable revenue stream from business or residential customers is now an option open to any individual who wishes to spend a little time researching the market. The proliferation of companies who offer support for Asterisk based business solutions and the use of Opensource software means that customized features are possible for each client of the ITSP without substantially increasing expenditure. An ITSP can be created on a global basis to meet the telephony needs of clients in a wide range of situations and by choosing a reliable provider of VoIP platform and carrier seamless integration into existing networks is a simple matter.

ICTBroadcast Version 3.0 Released , The best autodialer software in market

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We are pleased to announce release of ICTBroadcast Version 3.0 , ICTBroadcast is now a mature and professional product hence ultimate choice for telemarketers, carriers and Internet Telecom Service Providers (ITSP).

ICTBroadcast Version 3.0 include following enhancements

Improved Dashboard

ICTBroadcast 3.0 has an improved and more informative dashboard. Now admin can view server status, All active campaigns, and statistics of the logged-in users. A widget like view now enables to view the details by just clicking on it.

Innovative IVR Designer, create your custom IVR through drag and drop

IVR Designer now has 5 more applications. Now you can design your custom IVR with a lot more ease. New applications allow to speak custom values of Alphabet, Numbers, digits, date and time. Now improved Record application allows you to record complete IVR, a part of it or only the call recipient response. And these recordings can be downloaded or played in campaign summary with just a single click.

Implementation of Load Balancing

ICTBroadcast 3.0 now has built-in support for Load balancing. ICTBroadcast has been integrated with RabbitMQ that enables ICTBroadcast to scale and connect with multiple Asterisk servers and multiple Kannel nodes without any hassle.

Addition of Email Campaign

ICTBroadcast 3.0 supports creation of Email campaigns. Now you can choose your target medium from Voice, SMS, Fax and Email. No need to look a separate platform for Email campaign. Everything is bundled into a single software package. Buy one and get all.

Addition of Survey Campaign

Survey campaigns now has support for open-ended questions. Now you can create choice based questions that require key press from call recipient as well as open-ended questions that require call recipient to record their message. You can view the result of survey campaign in an informative chart report and can listen and download the recorded messages of the survey campaign.

Advance Campaign Management

Advanced campaign management allows to control the channel limits of the running campaign. Now you can monitor the status of your campaign and control it the way you want.

System wise configuration

System wise configuration allows to manage the behavior of the overall ICTBroadcast system. It allows to configure your campaigns, Forms, and Servers by advanced users.

Supporting variety of broadcasting scenarios

ICTBroadcast 3.0 supports a variety of broadcasting scenarios including but NOT limited to Voice, SMS, Fax and Email broadcasting, interactive voice broadcasting, survey, alert system, notification system, polls, telemarketing and custom IVR.

Support for Custom Variables / Tokens

Play custom message for each recipient using custom variables / token. Custom Tokens are supported in SMS messages, Email templates, IVR Applications like TTS, Say Alpha, Say Date, Say Time, Say Number and Say Digit.

Improved Campaign statistics and reporting

ICTBroadcast 3.0 has an improved campaign summary and reporting. With its chart based reporting, viewing and analyzing a campaign is much more easier than before. You can now view the contact details (contact list) of those who answered your call and of those where call Failed (due to Busy, No Answer, Congestion, Unknown error). You can export these contact lists for further processing.

REST based API's to integrate with third party software

ICTBroadcast 3.0 also comes with easy to use REST based API that can be used with any third party software or web interface. This can be used in a variety of ways from showing campaign statistics on the web page to creating new users through a function call.

Watch video for ICTBroadcast Introduction

Hotel Management System for Elastix

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[posted at Elastix.org] Two blog posts in 24 hours? We’re on a roll here!
Based off PIAF’s Hotel Management System, we’re going to make some minor adjustments to have this work with Elastix. This is a brilliant Hotel Room Management system with the ability to restrict the calls from Ext’s when they are not checked in, per-second billing, and more!

Contents:

1. Installing the base system
2. Updating the config
3. Fixing up HTTPS
4. Updating dial-plan
5. Closing words

1) Installing the base system

First you’re going to need to ssh in to your Elastix PBX System.
Then, run the following:

cd /var/www/html
wget http://www.kefa.it/hotel.tar.gz
tar zxvf hotel.tar.gz
rm hotel.tar.gz
cd hotel
mysql -uroot -peLaStIx.2oo7

Very similar to the default steps taken on the official documentation, except we’re going to use the Elastix SQL password rather than the script they’ve got there. This is also easier than the official doco, just one step instead of two.

2) Updating the config

We’re now going to modify the configuration file so that it works with Elastix due to the different Username / Passwords used between distributions.
Edit config.inc.php with your favorite editor. In my case I use nano:
nano -w config.inc.php

Now you’re going to modify the following three paragraphs so that they match below:

//Hotel database
$dbhost = "localhost";
$dbpass = "eLaStIx.2oo7";
$dbuser = "root";
$dbname = "Hotel";

//Asterisk database
$dbhost2 = "localhost";
$dbpass2 = "eLaStIx.2oo7";
$dbuser2 = "root";
$dbname2 = "asterisk";

//Asteriskcdrdb database
$dbhost3 = "localhost";
$dbpass3 = "eLaStIx.2oo7";
$dbuser3 = "root";
$dbname3 = "asteriskcdrdb";

Save and exit and you’re done with the config.

3) Fixing up HTTPS

When you visit the Elastix WebGUI, there’s a condition that rewrites the URL to include HTTPS. However, when this system tries to check the state of an Extension, it must do it via HTTP and not HTTPS, so we add an exception.

Edit /etc/httpd/conf.d/elastix.conf with your favorite editor and add these lines at the very bottom:

RewriteEngine Off

What this will do is add the /hotel directory as an exception so you don’t have to use HTTPS to get to it. We leave HTTPS on for the rest of the PBX, just because there’s no real reason for us to turn it off.
Now, restart apache with:

/etc/init.d/httpd restart

You should now be able to access your WebGUI without HTTPS at: http://pbxip/hotel/
If not, then go back and try again, as this part is crucial to the redirection to Reception when a phone is not Checked In

4) Updating dial-plan

Now we’re going to add in to the Dial-Plan and tell the PBX what calls can and cannot be made when the phone is not Checked-In.

Edit /etc/asterisk/extensions_custom.conf in your favorite editor and add the following underneath the [from-internal-custom] header:

;lets guests call other rooms without being checked in
exten => _XXX,1,Dial(SIP/${EXTEN})
exten => _XXX,n,Hangup()
;lets guests call freephonenumbers without being checked in
exten => _1800NXXXXXX,1,Goto(outbound-allroutes,${EXTEN},1)
exten => _1800NXXXXXX,n,Hangup()
exten => _X.,1,Set(result=${CURL(http://127.0.0.1/hotel/checkuser.php?Ext=${CALLERID(num)})})
exten => _X.,n,NoOp(Results: ${result})
exten => _X.,n,GotoIf($["${result}" = "OK"]?OK)
exten => _X.,n,GotoIf($["${result}" = "KO"]?KO)
exten => _X.,n(OK),Goto(outbound-allroutes,${EXTEN},1)
exten => _X.,n,Hangup()
;call front desk if not checked in
exten => _X.,n(KO),Dial(SIP/200) ; used for just one extension
;exten => _X.,n(KO),Goto(ext-group,0,1) ; used for ring groups
exten => _X.,n,Hangup()

If you want to allow additional free calling, just duplicate the lines with “1800? in them. Mines quite different for New Zealand, but I’ve left it as american to give you a good idea of what you can do there.
You’re also going to want to adjust the third-to-last line where it says “SIP/200?, replacing 200 with the Extension of Reception.
Or, you could just comment that line out, and uncomment the second-to-last line to use Ring Group 0 as your “Reception” extension. The choice is yours.
Save the file and run:

asterisk -rx "dialplan reload"

5) Closing words

From here, you’re on your own. It’s dead easy to log in and get started. You’ll first need to import all the rooms which it takes from the Ext Caller ID name, so it pays to give your Ext’s decent names like “Room 201? for Ext 201, just for example.
If you run into troubles, post here and myself or the Elastix team would be more than happy to assist.
If you get this successfully installed, just leave us a quick comment to say Hi and tell us you’ve done so, it’s always nice to know our time is appreciated.
Cheers

Chill.

Open Source Voip Applications in Telecom Industry

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Open source software has leading role in emerging of present Information and communication technologies (ICT), Following is list of top rated , mature and reliable open source applications that revolutionized the communication concepts.

IP Telephony applications

  • FreeSWITCH

    A Cross-platform, Scalable, Stable Multi-Protocol Soft Switch ,The project inspired from renown Open Source asterisk PBX system but very well designed and has a bright future
    http://www.freeswitch.org

  • Asterisk

    A software-based, Open Source Converged PBX system that has revolutionized the traditional PBX industry
    http://www.asterisk.org

  • Kamailio

    SIP proxy server, call router, and user agent registration server used in Voice over Internet Protocol and instant messaging applications.
    http://www.kamailio.org/w/

  • OpenSIPS

    SIP proxy server, call router, registration server and redirect server suitable for internet telephony service providers to offer SIP based telephony services to their customers.
    http://www.opensips.org

  • Sipexes

    SipX ECS (Enterprise Communications Server) is SIP based voice over IP telephony server, an ideal enterprise grade, SIP based communications system (IP PBX)
    http://www.sipfoundry.org

  • Gnugk

    H.323 gatekeeper based on the OpenH323 stack. A gatekeeper provides address translation, admissions control, call routing, authorization and accounting services
    http://www.gnugk.org

  • OpenSBC

    Session Border Controller and B2BUA featuring Registrations, B2BUA, NAT traversal and ENUM
    http://www.opensipstack.org

Asterisk & Freeswitch GUI interfaces

  • BlueBox

    Latest monitoring and management tool, full-featured PBX GUI of freeswitch
    http://www.2600hz.org

  • FreePBX

    A full-featured PBX GUI supporting both asterisk and freeswitch
    http://freepbx.org

  • Elastix

    An open source Unified Communications Server software that brings together IP PBX, email, IM, faxing and collaboration functionality
    http://www.elastix.org

  • TrixBox

    Web based asterisk PBX managment software utilizing best open source tools built over asterisk
    http://www.trixbox.org

  • PBXinaflash

    PBXinaflash is a web based asterisk PBX managment software built over asterisk like Elastix and Trixbox
    http://pbxinaflash.net/

  • Druid

    Druid is unified communications platform for enterprises. It allows companies to deploy high end IP communications services using off the shelf hardware and IP phones.
    http://www.voiceroute.org

  • Vicidial

    Vicidial ( Astguiclient ) is a professional web based inbound / outbound call center solution built over asterisk
    http://astguiclient.sourceforge.net

Communications Frameworks

    Plivo

  • Plivo is freeswitch based communication framework which provides web developers with basic building blocks to create voice based applications in the form of RESTXML and RESTAPI (REST-based APIs) also Plivo is compatible and works with Java, .NET, node.js, Perl, PHP, Python or Ruby programming languages.
    http://www.plivo.org

Communication Libraries

  • PJSIP

    Pjsip is sip based open source high performance communication library with multimedia capabilities written in C language for building embedded/non-embedded VoIP applications.
    http://www.pjsip.org/

Broadcasting Platform

ICTDialer

ICTDialler is next generation powerful auto dialler software that support sms messaging, fax broadcasting , voice broadcasting . It is bundled with IVR builder to create custom IVR with ease through web based drag and drop IVR Designer's tools .
http://www.ictdialer.org

Billing Systems

  • A2billing

    A2Billing combined with Asterisk is a physical Telecom Platform and Soft-Switch providing a wide range of telecoms services using both traditional telephone technology or VoIP. It contains a real time billing engine
    http://www.asterisk2billing.org

  • ASTPP

    A calling card voip solution built over asterisk and freeswitch
    http://www.astpp.org

  • Freeside

    Freeside is the premier open-source billing, CRM, trouble ticketing and provisioning automation software for wired and wireless ISPs, VoIP, hosting, service and content providers and other online businesses.
    http://www.sisd.com/freeside

  • CDRTool

    a rating and mediation engine for Call Detail Records generated by opensips and others
    http://www.ag-projects.com/CDRTool.html

  • ICTFAX

    An email to fax gateway, supports G.711 faxing , PSTN faxing and T.38 origination and termination .ICTFAX is complete faxing solution
    http://www.ictfax.org

  • CitrusDB

    A Multi-user Billing solution for internet service, subscriptions, consulting, and telecommunications , It Provides CRM, Ticketing, Invoicing, and Credit Card Batches
    http://www.citrusdb.org/

Voip softphones / Clients

  • Twinkle

    Twinkle is sip based softphone and support both voice over IP and and instant messaging communications , It is sip standard compliant with advance features including secure voice communications using ZRTP / SRTP.
    http://www.twinklephone.com//

  • QuteCom

    QuteCom is cross platform open source softphone supporting Voice, Video and IM with advance features also support SRTP, QuteCom is developed using Python , C and C++ .
    http://trac.qutecom.org/wiki

  • SFLphone

    SFLphone is SIP and IAX based standards-compliant enterprise softphone, for desktop and embedded systems. It support Multi line and TLS / SRTP encryption
    http://sflphone.org/

Mobile Clients

SIP Protocol Stacks and Libraries

  • eXosip

    eXosip is a library that hides the complexity of using the SIP protocol for mutlimedia session establishement. This protocol is mainly to be used by VoIP telephony applications
    http://savannah.nongnu.org/projects/exosip/

  • Juphoon SIP Stack

    Rich software SDK support SIP, SDP, XML, RTP/RTCP, HTTP, STUN, ABNF etc. Support Windows, Linux, ThreadX, Vxworks etc.
    libdissipate SIP stack
    http://www.juphoon.com/en/products/sip.htm

  • Libre

    Portable SIP Stack under BSD license with IPv4/v6 support (SIP,SDP,RTP/RTCP,STUN,TURN,ICE,DNS)
    http://www.creytiv.com/

  • MjSip

    MjSip is a complete java-based implementation of a SIP stack. It provides in the same time the API and implementation bound together into the MjSip packages.
    http://www.mjsip.org/

  • NIST SIP

    A series of SIP applications, tools and libraries in JAVA.
    http://snad.ncsl.nist.gov/proj/iptel/

  • Open Sip Stack

    MPL licensed SIP stack with ENUM, Presence (XMPP/SIMPLE) and NAT traversal. Reference implementation of Session Border Controller (OpenSBC) available.
    http://www.opensipstack.org/

  • PJSIP

    PJSIP is an Open Source Embedded SIP protocol stack written in C. The development of PJSIP is mainly focused on having small footprint, modular, and very portable SIP stack for embedded development purpose (although it's perfectly good for Win32/Linux/MacOS as well).
    http://www.pjsip.org/

  • Twisted

    Python protocol stacks and applications includes SIP support.
    http://www.twistedmatrix.com/

  • SIP SIMPLE client SDK

    High level middleware on top of SIP, RTP, MSRP and XCAP protocols
    http://sipsimpleclient.org/

  • ivrworx

    high level Lua interface to SIP/RTSP/MRCP, for testing distributed VoIP scenarios (windows, Vista+ clients).
    http://code.google.com/p/altalena/

Kamailio Elastix Installation and Integeration

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[posted at Elastix.org]

Part 1

WARNING: DO NOT INSTALL THIS ON A PRODUCTION MACHINE! Some of you may want be wanting to increase the capacity of your Elastix machine. Asterisk generally has issues when you start to put a lot of SIP peers on it. One of the best ways to get around this is to install a proper SIP server and join it with asterisk. This allows you to register all your SIP handsets to the SIP server and make calls through to your asterisk machine.

The Kamailio module for Elastix allows you to easily integrate Elastix with a enterprise SIP server. All SIP handsets register to Kamailio. All internal SIP calls stay within Kamailio however if a call is external then Kamailio will push the call out to asterisk to deal with it.
To install the Kamalio module you first need to create the directory for it to go in. It installs itself from the directory /var/kamailio

mkdir /var/kamailio

Now change to the directory

cd /var/kamailiol

Download the kamailio integration file from the MBIT website

wget http://www.mbit.com.au/kamailio-v2.tgzl

Untar the file into the directory

tar zxvf kamailio-integration.tgzl

In the directory now will be a heap of files you want to execute the script that installs all the modules for you. To run the script execute install_kamailio.sh

./install_kamailio.shl

Soon as it is installed we will move to the next part for configuring Kamailio. The installation can take a while so please be patient. The installation can only be used with Elastix 2.0.


Part 2

This article moves on from Part 1. I have updated the kamailio installation file so you will need to download this and run through Part 1 again. Make sure it all installs correctly.

The first step is to go into the /etc/kamailio.cfg file. In this file is the configuration for kamailio. When calls are between Kamailio extensions the call stays within Kamailio. If the call is to the PSTN then Kamailio sends it to asterisk. In the kamailio.cfg file we need to set a couple of parameters for your system. I have been building my test machine on the IP address 192.168.1.150. You will find 2 places where I have set this IP in the kamailio.cfg. Change these entries to the IP address of your machine using nano.

nano /etc/kamailio.cfgl

Then restart Kamailio

service kamailio restartl

Make sure kamailio is running

service kamailio statusl

The next step you need to do is go into freePBX. Under http://yourelastixip/admin is the freePBX interface. Go into module admin and you will find a new module called Kamailio integration. The Kamailio integration basically allows you to register handsets to Kamailio on port 5060. It also sends all external calls through asterisk when you make PSTN calls or voicemail calls. Once you have activated it go into the extensions tab in freePBX.

When adding a Kamailio extension you need to add a custom extension. If you add a SIP extension under freePBX this will simply be an asterisk extension. First add your extension number and display name. The last field required is the Kamailio password. You can scroll down and you will find an extra section called Kamailio services. Make sure it is enabled for the extension and you put in a password, click submit in the freePBX interface and reload freePBX.

The final step is to add a trunk in freePBX. Create a SIP trunk with the trunk name kamailio.
Under the peer details put

port=5060
allow=all
type=friend
context=from-internal
host=”yourelastixip”
canreinvite=nonat

This will allow calls to flow between kamailio and asterisk.
When you are done you should be able to register a phone to Kamailio with the details you put into freepbx.

Troubleshooting
If some of the packages dont install correctly try cleaning up yum

yum clean all l

Then run the script again.
If everything installed correctly and you still cant make calls from Kamailio to asterisk try copying the kamailio.cfg from /var/kamailio again to /etc/kamailio. Then edit the IP address that I detailed above. Once this is done restart kamailio and try calling.

ICTInvoice, A complete Multitenant Hosted PBX Solution for Telecom Service Providers

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ICTInvoice is open source Elastix PBX module for billing and invoice managements, it convert Elastix PBX into multi tenant Hosted PBX platform suitable to offer hosted PBX services to Small Business Owners and Enterprisers . It is open source GPL v3.0 software, developed and maintained by ICT Innovations

ICTInovice enhances capabilities of Elastix billing and empower elastix admin to automatically generate and email invoices to Elastix Users on monthly basis based applying rates already set also enabling users to view or download their invoices in PDF format from elastix web interface. The invoices are generated per extension basis as well as per user basis also ICTInvoice generate different summary reports of all Users to admin. ICTInvoice has multi tenant capabilities as system admin can manage multiple companies , multiple users per company and multiple extensions per users by assigning / deassigned from web interface, ICTInovice enable admin to manage multiple invoice formats for each company creating different templates respectively

Main Features

Multi tenant , multiple companies addition supported
Each company supporting multiple users with multiple extensions per user
Generate and email invoices automatically on monthly basis on particular date already set
Generate and Email Invoices manually through web interface
Elastix Rate Engine integrated for outbound billing
Inbound billing support
Outbound billing support
Summery reports per extension wise or user wise
Call Detail reports per extension wise or user wise
and much more

Installation Instructions

Now we will move forward to learn how to install and configure ICTInvoice

1. First of all we have to download Latest version of Elastix unified communication system and burn it as ISO and install and configure on our machine as per installation instruction provided by Elastix web site

2. Install and activate "Developer Module" (elastix-developer-2.2.0-3) on Elastix PBX System as per installation instructions. Don't install the latest version of Developer Module because there is a bug in this Developer Module (2.3.0-1).

3. You can install Developer Module by using this yum command (yum install elastix-developer-2.2.0-3)

4. Download latest version of ICTInvoice from sourceforge.net web sites http://sourceforge.net/projects/ictinvoice

5. Click System => Load Module and click on browse button to select already downloaded ICTInvoice package file and it will install ICTInvoice on PBX System

Admin Manual

User Managment

Create users with required privelages and role through elastix menu system => user managment

Extension Mangment

Create required extensions through elastix menu PBX => Extension batch

Create New Company Templates

Click Invoices => Invoice management => Company templates => create new company template

Assigning users to company

Click invoices => Invoice management => Company Users

Select company name from top

Check un selected users from list

Click on top button to assign / deassign users to company

Assigning extensions to Users

Click Invoices => Invoice mangment => User Extensions

Select user from top

Select un assigned extension from list

Click on top button to assign / deassign extension to users

Create Invoices

Click Invoices => invoice managment => Create inovices

Select company name and user name from top right

Click on button "create invoices" to create invoices

View & Download Invoices.

Click Invoices =? My Invoices

Click on view to view invoice or click on download button to download invoices in PDF format

Download ICTInvoice

Download from Sourceforge.net, Please click here http://sourceforge.net/projects/ictinvoice/


ICTFAX, open source Fax Over IP Solution

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ICTFAX, is multi tenant open source Fax Over IP solution for your faxing needs , ICTFAX is web based software solution for service providers based on open source Spandsp,Asterrisk , drupal CMS and Astbill with advance multi service prepaid billing capabilities and supporting G.711 pass through faxing, PSTN / T.30 based Faxing and T.38 based Fax over IP. It can be used in following scenarios for fax over IP implementations

Email to fax
web to fax
Fax to email

Please visit http://www.ictfax.org for more details

ICTDialer, An Open Source Unified Communication Autodialer

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ICT Innovations released a completely free and open source broadcasting solution. ICTDialer is open source Unified Communications autodialer and telemarketing software solution, ICTDialer powered with re-known open source Content Management system Drupal and Powerful l freeswitch based communication framework Plivo with multi-tenant and Voice, SMS & Fax broadcasting capabilities . It can be scaled to blast thousands of simultaneous calls . ICTDialer is capable to fit in various telemarketing and broadcasting and telemarketing scenarios. It empower user with capabilities of Drupal CMS and Plivo Communication Framework

Following are some screenshots of ICTDialer

You need to add broadcasting gateway to terminate your traffic , add , export your contacts in contact groups and add recordings before you run new campaigns

ICTFAX Version 2.0 beta , open source Fax Over IP Solution

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Pleased to release new version of ICTFAX V 2.0 beta based on Powerful communication framework Plivo and Re-known open source CMS Drupal 7.0 , ICTFAX Ver 2.0 beta is completely rewritten software based on latest and proven open source technologies ,

Previous Version of ICTFAX was based on Drupal 4.7 and was not compatible with new release of PHP therefore giving a lot of issues to users and only experienced users was able to cope with these issues also It was a tedious job to upgrade ICTFAX to new version of Drupal , We , at ICT Innovations worked on this projects for months and ported most previous code into Drupal 7.0 and rewritten communication part into Plivo Communication Framework therefore changing communication engine of ICTFAX from Asterisk to Freeswitch

Freeswitch is more stable and proven communication engine working at back-end of ICTFAX , making it more reliable and stable Fax Over IP solution

For more information, please visit http://www.ictfax.org

ICTDialer, An Open Source Unified Communication Autodialer

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ICT Innovations released a completely free and open source broadcasting solution. ICTDialer is open source Unified Communications autodialer and telemarketing software solution, ICTDialer powered with re-known open source Content Management system Drupal and Powerful l freeswitch based communication framework Plivo with multi-tenant and Voice, SMS & Fax broadcasting capabilities . It can be scaled to blast thousands of simultaneous calls . ICTDialer is capable to fit in various telemarketing and broadcasting and telemarketing scenarios. It empower user with capabilities of Drupal CMS and Plivo Communication Framework

Following are some screenshots of ICTDialer

You need to add broadcasting gateway to terminate your traffic , add , export your contacts in contact groups and add recordings before you run new campaigns

ICTFax Version 3.0 , A Fax over IP Solution , released now

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We are pleased to announce that ICTFAX Version 3.0 is released. New release completely removes Plivo Framework from ICTFAX. Now ICTFAX no longer depends on Plivo for communication with FreeSWITCH. Instead, ICTCore has been introduced as a new lightweight communication library. ICTFAX uses ICTCore to communicate with FreeSWITCH. Apart from this major change, other features that are included in this release includes GUI based trunk provider configuration, multiple trunks allowed, documentation converted to markdown syntax, attachement file name with spaces issue has been fixed, error on retry issue fixed along with other minor bugs.

ICTCore Communication Library

ICTFAX now uses ICTCore for communication with FreeSWITCH. ICTCore is specifically built to provide reliable communication between ICTFAX and FreeSWITCH. It is developed using the object-oriented principles. Previously, Plivo framework was unable to cope with communication issues and was unable to provide a reliable connection between ICTFAX and FreeSWITCH.

Main reason to abandon Plivo framework from ICTFAX was that Plivo was not specifically built for faxing and we were using it with adhoc modification to work it for ICTFAX. That solution worked initially but later, due to no real intention from the open source community of Plivo to include fax support in it, we realized that there sould be a communication library with built-in fax support along with voice communication for FreeSWITCH. Hence, ICTCore is the solution with voice and fax communication support for FreeSWITCH. It provides easy to use API for developers to build complex fax and voice applications.

Additional Features

Other ICTFAX features that includes in this release is GUI based trunk provider configuration. In older versions, trunk configuration was done in two steps. First, an xml file was created in FreeSWTICH profiles directory manually. Then in second step, a new Trunk provider entry was created for this xml file in the GUI. Now, only one step is enough to configure trunk. No xml file is created. Just enter the trunk settings in GUI and xml file is created automatically.

Documentation has also been converted to markdown syntax that is easy to manage. Attachement file name with spaces issue has also been fixed. An old issue with retry is also fixed along with other minor bugs in new release.

ICTFAX: http://www.ictfax.org
Download Link (Github): https://github.com/ictinnovations/ictfax
Download Link (Sourceforge): https://sourceforge.net/projects/ictfax/

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