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  • 09/05/17--13:05: Voiceprint
  • Voiceprint

    Read your voicemail in your inbox with Voiceprint. It's the easiest way to save time checking business voicemail.

    • Pricing
      • Simple, easy to understand invoice
      • $3 per email address per month
      • $.05 per voicemail transcription
      • Provider discounts and whitelabel available

    • Simple Setup
      • No software to install
      • No 3rd party scripts
      • Completely email based

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  • 09/05/17--13:06: TheMediaTown
  • TheMediaTown is a fast growing bulk SMS service provider company in Mohali, India. They have various packages for promotional bulk SMS, transactional bulk sms, and reseller bulk SMS service.
    They provide Bulk SMS service at a very low price. They have dedicated support team to help our customers.

    They also prove reseller bulk SMS gateway API for who want to start their own business in SMS marketing. You can access reseller panel after getting the reseller bulk SMS package and start to selling SMS packages at your on cost, it means you can customize the cost according to you and your potential customers.

    The Media Town
    F-135, Industrial Area
    Phase 8B, Mohali. Punjab

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  • 09/05/17--16:45: softswitch
  • Softswitch is a central device in a telecommunications network which connects telephone calls from one phone line to another, across a telecommunication network or the public Internet, entirely by means of software running on a general-purpose system. Most landline calls are routed by purpose-built electronic hardware however, soft switches using general purpose servers and VoIP technology are becoming more popular.
    Nowadays, many telecommunications networks make use of combinations of softswitches and more traditional purpose-built hardware.

    A softswitch is also a VoIP server, providing a soft switch platform with full IP PBX call features. The most difference from IP PBX is its enormous numbers of users.

    Typical application networking diagram

    Typical softswitch application.jpg

    See also

    Softswitch Manufacturers and Providers

    Class 4 SoftSwitch Solution– Tandem Capabilities with Affordable Services and Support

    With Ecosmob Technologies, one of the leading Class 4 solution providers is focused on developing scalable, high-quality and cost-efficient VoIP softswitch solutions for telecom industries worldwide. Class 4 SoftSwitch development service offered by Ecosmob enables effective scaling of voice services while catering a time-tested point-of-presence between two carriers.

    Class 4 SoftSWITCH Key Features At Glance

    - Intelligent Call Routing
    - Geographic, Nongeographic & Nomadic Routing
    - User friendly Web Interface
    - Load Balancing & Failover
    - WLR Customers

    VoIP Class 4 SoftSWITCH Solution

    Our VoIP Softswitch has been mainly designed for routing local and long distance calls, thus the software is affordable yet advanced that meets the needs of telecom industries. We develop class 4 switch based on the protocol of a particular country. We integrates Authorization, Accountancy, Administration and Billing. So, avail with a software that bridges network management and monitors tasks.

    Key Benefits of Class 4 VoIP SoftSWITCH

    - Provides flexibility, reliability, security and scalability
    - Offers quick support
    - Provides advanced switching solution deployment
    - Offers high voice quality

    • Singapore based VoIP Service Provider INAANI presents iTel Switch Plus. This Softswitch is customizable according to your requirements which can be get in a package as monthly rental basis. The VoIP package contains a switch, a dialer along with the tunnelling software. The rate of the package varies by the number of concurrent calls. The softswitch along with the package is a complete set for your VoIP business.

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  • 09/06/17--07:31: Old News
  • This page lists all the old VoIP news stories from the home page.

    August 2017

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  • 09/06/17--11:11: Hosted PBX
  • What is Hosted PBX?

    Unlike a traditional PBX, or Private Branch Exchange, which requires a large investment and ongoing maintenance and training, a hosted PBX is a cloud-based PBX system accessible via an IP network. Rather than being responsible for hardware, software, training, maintenance, and more, a hosted PBX provider takes care of it all. In addition to being completely managed off-site, resulting in no IT or installation costs, a hosted PBX system also provides businesses with the ability to manage their phone systems via a user-friendly control panel. For these reasons and more, hosted PBX systems are becoming increasingly popular solutions for today’s growing small to medium-sized business owners.

    Features and Benefits of Hosted PBX

    If you own a business with fewer than 300 lines, hosted PBX can provide you with a multitude of features and benefits. For starters, a hosted PBX system is much less expensive than a traditional on-site system. With hosted PBX, there is no need to buy expensive hardware and software, pay for installation, and manage the system. You simply pay a monthly fee and enjoy all of the benefits of the service.

    While less costly than standard on-site PBX systems, hosted PBX solutions are teeming with valuable features, such as on-hold music, call waiting, call routing, transfers, and more. Moreover, as the popularity of hosted PBX continues to grow, additional features like auto attendants, extension dialing, and ACD queues are being introduced.

    A hosted PBX can also be deployed immediately. In fact, most business owners are able to have their hosted PBX setup and running in under a day. Best of all, newer and more flexible hosted PBX applications are being deployed as well. Adding these applications and features can typically be done with a simple download or click of the mouse.

    The ability to add new features as they roll out is just one example of the scalability of hosted PBX. Additional lines, phones, and even an entire new department can effortlessly be added, which would be much more complex with a traditional on-site PBX.

    When you opt for a hosted PBX and outsource all the techie stuff, you will find your stress levels lowering as well. With a hosted PBX, a mountain of responsibility will be taken off your shoulders, allowing you to focus your attention on more important matters, like improving your bottom line.

    Hosted PBX Tips and Considerations

    Hosted PBX systems are incredibly easy to setup and utilize, but there are a few things you need to do when opting for one of these modern business phone solutions, such as:

    • Make Phone Number Arrangements – A temporary phone number is necessary for the porting process, but if you already have an existing number to port over, you must remain with your current provider until the number is ported to your new PBX provider. If you are getting a new number, it’s important to update others and let them know of the change through an email blast or similar means. While brief, all calls can be conveniently routed to cell phones during the transition period. Just remember, if you cancel your service with your previous carrier prior to porting your number, the number will no longer be yours to keep.
    • Arrange Your Dial Plan – When switching over to a hosted PBX, organizing the routing of your calls is one of the first things that must be done. In order to do so, you’ll need to define various rules for your calls, including:
      • The buttons used to activate voicemail and other features
      • The hours of operation for your desk phones
      • Setting up your directory
      • The handling of faxes
      • Sequential and simultaneous ringing
      • How off-hour calls are handled
      • Configure your emergency 911 settings

    Review Your Bill

    As you can see, setting up a hosted PBX is remarkably easy. However, after doing so, you should review the bills for the first and second month to ensure you’re receiving and utilizing all of the services you’re paying for. Depending on which company you choose as your hosted PBX provider, the first month’s bill may or may not reflect activation fees, setup fees, and number porting fees. The second month’s bill should be a typical bill without added charges for the setup process. By this point, you'll be well on your way to enjoying one of the most streamlined and hassle-free business phone systems available for small and medium-sized business owners. ...

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  • 09/06/17--11:17: VoIP Origination
  • Please add information to this page about VoIP Origination.

    What is VoIP Call Origination?

    One of the terms most often used when talking about VoIP communications is call origination, which basically has to do with how a telephone call starts and how it travels to the receiver once it does. The following will provide an in-depth look at what call origination is, how it works and what type of hardware is required.

    What is Call Origination?

    VoIP stands for "Voice Over Internet Protocol". This means that phone calls utilize a technology that allows the calls to be sent directly over internet networks, which is a much cheaper way to make calls. Call origination refers to the point where the call starts, which takes place over the PSTN telephone network and transferred to their destination through the internet. It's important to note that a phone call through VoIP starts between the initiation point and the destination point, which are referred to as the originator and terminator respectively. There are typically different types of call origination depending on the services provided by the ITSP involved. The only way for call origination to work is if the VoIP originator has a call termination arrangement with a VoIP terminator.

    Required Hardware

    The best aspect of VoIP services is that there are hardly any noteworthy hardware requirements. All that is necessary is a gateway, which essentially transfers phone calls from the internet and onto PSTN lines. Since a gateway must interact with both the internet and standard PSTN lines, there are two interfaces necessary for a gateway, including a telephony interface that takes digital and analog lines and an Ethernet interface as a connection between the gateway and the internet. It's important to understand that a digital line can support a large amount of calls at one time, which can range anywhere from 20 to 30 depending on the type of line that is chosen. In comparison to a digital line, an analog line can only support one phone call at a time.

    How Call Origination Works

    Call origination is a fairly simple process that is a bit more complicated to explain. In essence, the gatekeeper mentioned previously will receive the calls and requests from the dialer. When a user makes a connection to the dialer, the gatekeeper will ask a Radius server to check if the user has input the correct password and username. The Radius server will then answer the gatekeeper with a yes or a no. If a yes answer is received from the Radius server, the user the has the ability to make a phone call from the origination point. Once the number has been placed, the gatekeeper once again receives a request for the phone call to be made. It is at this point where the gatekeeper will interact with the Radius server again to see if the user in question has enough money to make the phone call. The Radius server will then connect to the billing server to ascertain how much money has been provided by the user for this specific call, in order to nail down how long the call can last before being cut off.

    The billing server will then take a look at the location at which the call is originating from in order to measure the current rate of pay, which all depends on whether the call is being made in off peak or peak hours. Once this has been determined, the relevant information will then be sent back to the Radius server. Once the Radius server has received this information, it will be sent back to the gatekeeper. The user is then provided with the IP address for the gateway at the destination to which the call is being placed. This gateway will send the call to the final destination. If no other issue arises during this time, the call can take place and the two users can talk for however long the duration of the call is. If the call takes place once the maximum time allotment has been reached for the call, it will be disconnected and more money will need to be provided in order to make another call and continue the conversation. When the call has come to a conclusion, whether by the callers themselves or because they were disconnected, the initial user will have the total price of the call deducted from their payment source by the billing server.

    Types of VoIP services

    There are two basic types of VoIP services, including PC to phone and phone to phone. The provider for the PC to phone service will place a dialer on the internet that the user of the service can download and install. This dialer allows for an account to be created with any type of payment source that is allowed by the service provider. Once this is done, the user can make and receive calls. ...

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  • 09/07/17--08:48: DID Service Providers
  • A Direct Inward Dialing service provider delivers a telephone number over VoIP protocol including SIP, IAX2 or h323. This DID Phone Number will receive calls using a sip soft phone, hard phone, or an IP PBX. The charges are charged per month, and per min, and per channel based. You do not require a hardware card in case you have a DID service provider to receive call, ie a PRI Card or a Analog Card. The call travels to you all the way over the Internet

    SMS enabled DID Providers

    • AAA+ Rated Freevoice DID provider. No Setup fee’s, Commitments or Terms. Origination DID’s Available in all US Area codes. SMS, E-911 and CLID features. Fax T38 support. Toll Free numbers, Live US Support. Call 24/7 to 800-834-8999
    • Provides Lowest Rental DID's,Unlimited Channel DID's,SMS supported DID's.Wholesale Pricing from First DID.
    • DIDWW.jpg
      DIDWW - the source for wholesale International DIDs and Toll Free Virtual Numbers. We provide Voice and SMS enabled DIDs in many countries. SMPP, HTTP and Email SMS forwarding.
    • MultiTEL is providing retail and wholesale Worldwide DIDs - over 90 countries. Pick your own SMS enabled DIDs from over 40 countries. (US, UK, CA, Germany, etc). Free access to JSON and SOAP API provided along with programming samples. No contracts, pay per month, instant activation. SMS forwarded via e-mail , URL or as SMS/Text message. All calls are forwarded to SIP, PSTN or to our free Hosted PBX. Coverage and numbers always available in stock from more than 90 countries.

    DID Providers by country


    • MultiTEL is providing retail and wholesale Angola (Luanda) DIDs . Free access to JSON and SOAP API provided along with programming samples. No contracts, pay per month, instant activation. Payment by card, paypal or bank transfer. All calls are forwarded to SIP, PSTN or to our free|Hosted PBX] . Coverage and numbers always available in stock from more than 90 countries.
    • TeleCallMart Local and Toll Free Numbers from 1$, Voip calls, SIP Phone, Auto Attendant, DTMF, REST API. No monthly fees and 0$ setup.


    • DIDNumberHub.comLowest Rental Argentina DID /Virtual Phone Numbers/DDI Numbers @$3/month including free PBX. with the Free forwarding via SIP/IAX2 ,Asterisk or VOIP. Phone Numbers all over the world are available. Free PBX __. Toll Free Number Available without Monthly commitments.Lowest Rentals from 1st DID.Unlimited Channel DID's available without per minute charge.

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  • 09/08/17--08:22:
  • About is the retail website of North America's leading VoIP solutions provider, VoIP Supply. offers a fast and easy way to find the VoIP hardware you are looking for. has products for consumers, small medium businesses, enterprises, government/educational entities, resellers and service providers. You can visit Here.

    VoIP Supply recently introduces our new Partner Portal for our registered resellers! You will find important documents, eductional resources, marketing materials and much more!

    Access to the portal here

    About VoIP Supply

    VoIP Supply is North America's leading VoIP solutions provider. Since 2004, VoIP Supply has delivered valuable solutions for some 60,000 customers worldwide.

    With over 30 passionate employees, 2,500 products, 40,000 square feet of office space and an unlimited number of VoIP solutions to meet your needs VoIP Supply has everything you need for VoIP, whether you are a consumer, business, service provider or reseller. Offerings specializes in standards based VoIP equipment and VoIP systems. In addition to complete VoIP systems, VoIP Supply offers:

    Besides VoIP equipment and complete Business Phone Systems, VoIP Supply offers a number of value-added services to consumers, businesses, resellers and service providers:

    • Full online catalog at
    • Network assessments
    • Solution design
    • Device configuration, set-up, and installation
    • On-going maintenance and support
    • Device fulfillment (privately branded)
    • Extended no-questions asked warranties

    Upcoming Events

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  • 09/10/17--03:38: Aastra 6730i
  • Aastra 6730i 3-Line IP Phone with LCD Display


    The Aastra 6730i offers exceptional features and flexibility in an enterprise grade IP telephone. With a sleek, elegant design and a compact footprint, this multi-line SIP telephone delivers the advanced features and performance traditionally found only in higher priced products.

    Featuring a 3 line LCD display, the 6730i supports up to 6 lines with call appearances, offers advanced XML capability to access custom applications and is fully interoperable with leading IP-PBX platforms.

    Supported by a host of Aastra configuration options and product enhancements via software releases, the value offered by the 6730i makes it is ideally suited for light telephone use for the small and home-based business applications.

    Key Features and Benefits

    Aastra Hi-Q™ Wideband Audio Technology

    All 6700i Series IP Telephones integrate Aastra Hi-Q™ wideband audio technology to deliver enhanced performance and voice clarity. Aastra Hi-Q significantly improves the audio quality of calls offering a truly superior voice experience on each audio path – handset, hands-free speakerphone or headset port. It makes conversations more life-like giving a richer user experience and increasing productivity.

    XML Support

    The 6730i is equipped with XML browser capabilities allowing access to customized services and applications. This allows creation of internal service applications using development guides available from Aastra. This feature provides unlimited potential to customize the 6730i to meet your specific business needs or vertical and CTI applications using the display and keypad.

    Enhanced Call Management

    With extensive storage capacity for personal directories, callers logs and redial lists, the Aastra 6730i can improve efficiency by providing more call information with the push of a button. This includes features such as shared call and bridged line appearances, call forward, call transfer, call waiting, intercom and 3-way conference providing enhanced call flexibility and control.

    Simplified Deployment

    From initial deployment and configuration to future enhancements and upgrades, the Aastra 6730i is designed to save your business time and money. Easily created configuration files, using any text editor and a variety of communication protocols, can be used to configure phones individually or centrally.


    Feature Keys

    • Up to 6 lines call appearances with LEDs. 2 dedicated keys
    • 4 navigational keys
    • Up to 8 programmable keys with LEDs. ...

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  • 09/10/17--10:59: OneCloud Networks Limited
  • OneCloud Networks Limited gives Hosted Cloud PBX, Unified Communications, VoIP Service, Team Collaboration and Contact Center Solutions in the UK that can enhance your business execution without expanding your expenses.

    We give far reaching UCaaS answers for organizations that incorporate talk, sound and video conferencing, desktop sharing, web meeting, full versatility usefulness, and the capacity to chip away at any gadget, anyplace, whenever all with a similar business character.

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  • 09/11/17--06:51: ENUM
  • ENUM - The bridge between the switched telephony network and the Internet

    ENUM (E.164 Number to URI Mapping) translates telephone numbers into Internet addresses. You can dial a telephone number and reach a SIP, H.323 or any other Internet Telephony user. This all happens in the background; you do need to do anything special while calling someone.

    A server with ENUM support will lookup a dialled telephone number in the ENUM tree of the DNS to see if there's alternate ways to set up the call instead of just calling out on the PSTN telephone line. ENUM may contain a reference to a SIP URI, a telephone number to dial, a web page or an e-mail address.

    ENUM is already supported by SIP Proxies like SER, Kamailio, OpenSIPS or SNOM 4S, VoIP gateways like Asterisk, Swyx, and SIP phones (SNOM).

    Enum uses DNS NAPTR resource records.

    ENUM RFC 6116 is a protocol developed by the IETF that uses the Internet DNS system to translate E.164 (i.e. ordinary) telephone numbers into IP addressing schemes (like SIP, H323 or Email). To register new Enumservices (or update existing ones) with IANA, RFC 6117 contains all the information you need.


    • ENUMER - Blockchain-based decentralized ENUM technology and implementation is released (11-Sep-2017).
    • New ENUM Standards Published (14. March 2011): RFC 6116, RFC 6117 and RFC 6118 update various older RFCs on ENUM.
    • is now a productive service (Spring 2011)., a service of the National Research and Education Networks (NRENs), can be used in the countries where the Golden Tree (e164. ...

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  • 09/11/17--10:11: VoIP Providers USA
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  • 09/12/17--22:16: T.38
  • T.38 is an ITU standard for sending FAX across IP networks in a real-time mode.
    FAX messages are sent as UDP or TCP/IP packets.

    • The IETFRFCRFC 3362 implements a media type called image/t38 for T.38 faxes.

    From RFC 3362:

    ITU-T Recommendation T.38 T.38 describes the technical features necessary to transfer facsimile documents in real-time between two standard Group 3 facsimile terminals over the Internet or other networks using IP protocols. The Recommendation allows the use of either TCP or UDP depending on the service environment.

    ITU-T Recommendation T.38 T.38 Annex D describes system level requirements and procedures for internet aware facsimile implementations and internet aware facsimile gateways conforming to ITU-T T.38 to establish calls with other ITU-T T.38 implementations using the procedures defined in IETF RFC 2543 SIP-99 and IETF RFC 2327 SDP.

    Note that ITU-T T.38 Recommendation T.38 (04/02) T.38 is an aggregation of the original ITU-T Recommendation T.38 (06/98)T.38-98 and all of the subsequent Amendments and Corrigendum including T.38D-00. While T.38 and T.38D-00 describe SIP procedures per SIP-99, the procedures can also be applied to the revised Session Initiation Protocol specification SIP.

    The Importance of T.30 ECM Error Correction Mode (ECM) in T.38 Deployments

    One of the most important features of the traditional facsimile standard (T.30) is ECM error correction. The inventors of fax recognized that the audio quality of PSTN phone lines could not be trusted to be 100% error free - it was not uncommon to hear static, or the occasional pop or crackle on some fax calls. ECM error correction allows the sending and receiving terminals to compare notes at the end of each page, and selectively retransmit any data that was not received the first time around. This retransmission process is continued until the received page is certified error free, and the transmission of the next page begins.

    Remarkably, many T.38 implementations, including those of top tier carriers such as Level3, XO, Verizon etc have explicitly disabled ECM error correction. This problem is discussed in detail here. Don't make this mistake! Be sure to enable ECM and insist your provider or PBX vendor do the same!

    Want to check to make sure you have ECM enabled? This handy ECM self-test tool can tell you in minutes.

    T. ...

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  • 09/12/17--22:17: Asterisk fax
  • Asterisk and fax calls

    Fax over IP

    Across the Internet even a G.711 codec fax transmission is unpredictable. An excellent discussion of why faxing and modems don't work well over VoIP can be found here. However, people often get perfectly good results on lightly loaded LANs. It still isn't perfect, as a burst of data on the LAN can still upset things, but some people get results they can live with.

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  • 09/14/17--05:58: WebRTC
  • Synopsis

    The practical implementation of VoIP was started on hardware based IP Phones. The idea was well received and was transferred into the concept of Soft Phones or software based IP Phones. These softwares always required some additional installation to the native Operating System. Most common examples of Softphones or Software based SIP client is Counterpath's X-Lite and Bria.

    The Evolution of Software Development made it possible to translate or formulate equivalent of almost every desktop based application to web based application. This brought major shift in Software Industry as the web browsers are integral part of almost every Operating System. SIP clients, were also transformed into Web Extensions. Most of the time, Flash was used to develop such extensions however, it always required extra plugin installation, thus decreasing system performance, and increasing chance to troubleshoot as it required additional resources to be deployed. And this problem gave rise to the concept of WebRTC.



    WebRTC provides the functionality of realtime multimedia applications without any installation of additional plugins, downloads or extensions. The ideal form of WebRTC describes such web based Real Time Communication independent of Browser being used by user. It's a Javascript based API originally being developed to develop browser to browser communication applications for Voice, Video and Peer to Peer File Sharing tasks.


    The architecture of WebRTC, as described by W3C looks something like this:
    WebRTCpublicdiagramforwebsite (2).png


    Major components of WebRTC include:

    • getUserMedia, which allows a web browser to access the camera and microphone
    • PeerConnection, which sets up the audio/video calls
    • DataChannels, which allow browsers to share data via peer-to-peer


    Chrome WebRTC Development Team

    Discussion List:
    Google Plus Page:
    Chrome WebRTC Issue Tracker: ...

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  • 09/15/17--21:32: Thirdlane Business PBX

  • 08/03/2017 - New version of Thirdlane Connect with support for voice, video, private and group chat, integrations, screen and file sharing and the new version of Thirdlane Multi Tenant PBX platform are available.

    Thirdlane Business PBX is a highly-reliable, cost-effective IP PBX and unified communications software platform that forms the heart of a versatile Voice over IP (VoIP) enabled telephony system. It provides small businesses, larger companies, and multi-site enterprises a flexible combination of the best of open source and commercially developed solutions, offering an alternative to high-cost telephony systems.

    In today’s fast-paced business environment, a reliable integrated communications system is critical to success. Thirdlane’s Business PBX delivers by including all the standard telephone system features you expect, plus advanced unified communications capabilities such as integration with email, messaging, and mobile platforms at no extra cost. Thirdlane systems have been field-proven for over ten years across thousands of customers worldwide, and are widely regarded for their unique combination of diverse features, flexibility, and bulletproof reliability.

    Thirdlane provides system administrators with all the tools required for easily making changes and adding users or devices. A unique deep customization ability allows each user or group their own feature set to meet virtually every business need. Thirdlane’s Business PBX also boasts an expandable open architecture and an extensive API (application programming interface) to allow integration with third-party applications such as CRM, ERP, accounting systems, and other business tools. The result is that your company can benefit from an easy-to-use yet highly adaptable set of advanced features with a low total cost of ownership.

    Key Features:

    • Advanced Call Features: Among the PBX features included are: IVR (auto attendant), conference bridges, call forwarding, transfer, call screening, call parking, call presence, ring groups, hunt groups, find me/follow me, call queues (ACD), direct dial (DID), fax handling, selective call screening and blocking, call recording, intercom, paging, voicemail to email, and much, much more.
    • Auto-Provisioning: Thirdlane Business PBX includes templates for auto-provisioning of devices such as Aastra, Cisco, Linksys, Polycom, Snom and Yealink phones and ATAs. This allows you to easily add phones and devices, individually or in bulk. Entire groups of users can be easily added and provisioned. Templates can be readily customized or added to support new devices.
    • Fine-grained permissions: Allow your administrators to easily control each user’s call permissions and features as appropriate for your business requirements, and to configure dialing rules on a per-user or per-route basis for operational economy and flexibility.
    • Highly Customizable: Choose from a number of Communications Manager and user portal GUI themes, and even customize menus and configuration files. Select from 12 supported languages, or add your own language translations or voice prompts. Easily add custom scripts to support user-requested features and integrate with most third party program’s API.
    • Proven Industry-Standard Components: Thirdlane systems are built with standard, proven components, including the Asterisk® telephony engine and CentOS® Linux, without custom patches. This allows you to easily update them if needed. Thirdlane also supports versions of Digium®-certified Asterisk and Red Hat® Linux for additional peace of mind in critical applications.
    • System Control and Updating: Unlike other solutions, Thirdlane doesn’t deny you root access to control your server for troubleshooting, integrating with third-party components, and performing system maintenance and customization. Software updates are managed by a Thirdlane repository and are easy to install, to keep your system updated with the latest security features. ...

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  • 09/15/17--21:34: Thirdlane Multi-Tenant PBX

  • Thirdlane Multi Tenant PBX is a reliable, feature rich, scalable IP PBX and unified communications software platform capable of hosting multiple virtual PBXs on a single server or a cluster of servers. It enables Internet Telephony Service Providers (ITSPs), telephony resellers, and VoIP entrepreneurs to offer a fully-featured hosted PBX to their customers, providing them an economical alternative to premise-based PBX hardware.

    Thirdlane Multi-Tenant PBX provides an easy-to-deploy, hosting-ready solution. The software is provided as a complete package, ready to install with all the components needed to build a powerful PBX hosting platform. With Thirdlane’s field-proven technology, you can offer great value to your customers by providing reliable hosting services and a customizable set of advanced features at a very competitive per-user cost compared to equivalent solutions.

    Thirdlane also provides system administrators tools for easily provisioning and managing PBX tenants, and the ability to provide each customer their own customized feature set to meet virtually every need. Thirdlane Multi Tenant PBX boasts an open architecture and an extensive API (application programming interface) to provide interoperability and ease of integrating with third-party components such as service provider portals, billing systems, CRMs, and other applications.

    Thirdlane Connect adds unified communications including private and group chat, voice and video conferencing, integrations, screen and file sharing to the latest version of Thirdlane Multi Tenant PBX platform.

    Thirdlane Connect is available on Google Play.

    Key Features:

    • High- Availability: Redundancy, geo-redundancy, automatic failover, and clustering capabilities are built into the core of all Thirdlane Multi Tenant PBX solutions. The high availability features can be configured during installation or added later to address your requirements
    • Auto-Provisioning: Thirdlane PBX is available with templates for auto-provisioning of devices such as Cisco,
    Polycom, Snom, VTech, and Yealink phones and ATAs. This allows you and your customers to easily add phones and devices, individually or in bulk. Templates can be customized or added to support new devices.
    • Fine-grained permissions: Allow your administrators or customers to easily control each user’s call permissions per user or per group as appropriate, and to configure dialing rules on a per-route basis for operational economy and flexibility.
    • Customizable for your Company and Users: Upload your company name and logo to rebrand the PBX, select from Communications Manager GUI themes and languages, add language translations, and even customize GUI menus and configuration files. Since scripts are built in as part of the Thirdlane architecture, you can easily add custom scripts to support customer-requested features.
    • Retain Full Control of your Server: Unlike other solutions, Thirdlane doesn’t deny you root access to your server for troubleshooting, integrating with third-party components, and performing system maintenance and customization. Software updates are managed by a Thirdlane repository and are easy to install, to keep your system updated with the latest security features.
    • Industry-Standard Components: Thirdlane employs CentOS® Linux, and doesn’t require custom patches. This allows you to easily update components of your system if needed. ...

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  • 09/18/17--12:21: Free Virtual PBX
  • Please list only FREE VIRTUAL PBX service providers and platforms here

    Multitel Hosted PBX
    Offering Hosted PBX services with termination to US/Canada at $0.0095/minute. International calling also available. We also provide DIDs from over 100 countries. Hosted on geographically diverse colocations (locations in North America, South America, Europe, Africa, Asia, Australia) - so you're always just 40-50ms away from our closest PoP - therefore you'll have great audio quality

    Try it now for free - pay only for your minutes or DIDs - 20 extensions / 20 voicemails / 20 IVRs available for FREE FOREVER :)

    Also check out virtual PBX reviews and more at

    See also

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  • 09/19/17--09:15: ICTFAX
  • Release Note

    Released new version of ICTCore V 0.7.5 on March 12, 2017 with following improments

    • Twig based template added for gateway configurations and Application data
    • Data and Token libraries updated
    • Sip, SMTP and SMPP added as provider sub-type
    • Multi tasking support improved for Task and Schedule
    • Namespaces and PSR-4 based auto-loading support added
    • PhpUnit support added for unit testing

    Released ICTFax Version 3.7.4
    Reported issues fixed, successfully tested both inbound and outbound faxing December 2016
    Released new version of ICTFAX Ver 3.2 on April 15, 2015 based on ICTCore, a new communicatiosn framework, fixing reported issues

    Released new version of ICTFAX Ver 3.0 on Nov 28, 2014 , completly rebased on ICTCore after dropping plivo , Old version of ICTFAX was based on Plivo and has several issue during installation
    Released new version of ICTFax ICTFAX Version 2.2.0 on Feb 13, 2014 , Fax over IP software implementation based on T.38 protocol also support G.711 pass through faxing and PSTN faxing



    ICT FAX is an open source business solution especially for faxing along with support of SMS and Voip with advance web based billing capabilities featuring TIME, Per PAGE and Per SMS based Billing , It supports G.711 , T.38 and PSTN faxing .ICTFAX is complete faxing solution and does not need to be integrated with other open source projects to function properly that makes ICTFAX a unique and innovative faxing solution.

    ICTFAX, a Faxing solution

    ICTFAX can be used in following faxing scenarios

    ICTFAX, a SMS solution

    ICTFAX can be used in following SMS sending scenarios

    • Email to SMS
    • Web to SMS



    Download open source Online FAX solution


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  • 09/19/17--18:24: ICTBroadcast
  • ICTBroadcast is multi tenant unified communication auto dialer software . It features Voice broadcasting, SMS messaging, Email marketing and Fax blasting , ICTBroadcast is being offered
    in two type of editions, ICTBroadcast Enterprise Edition suitable for Entrepreneurs and ICTBroadcast Service Provider Edition suitable for Internet Telephony Service Providers (ITSP). ICTBroadcast supports renknown open source Communication Engine Asterisk also it support open source SMS gateway named Kannel. ICT Broadcast is integrated with RabbitMQ to achieve scale-ability that enable it to blast thousands of concurrent either voice or fax calls using either VoIP ( SIP or IAX ) , PSTN and FOIP (T.38 / G.711 pass through ) .It is simple, reliable and user friendly web portal.

    ICT Broadcast platform support following type of campaigns

    Simple Voice Broadcasting )
    Voice Broadcasting with direct forwarding to Live agents support on answer)
    Interactive Voice Broadcasting / press 1 campaign )
    Survey / Polls )
    Inbound IVR campaigns)
    SMS Broadcasting )
    Fax Broadcasting )
    Email Marketing )
    Custom IVR voice broadcasting)

    New features added on december 2016

    CentOs 7 and Asterisk 13 support added
    Responsive theme added in ICTBroadcast
    Fail 2 ban support added for asterisk and ssh
    inbound ivr campaign support (inbound cost need to increase to USD 600)
    Conditinal survey support added plus no. of question in survey increased
    Option to download recordings
    DNC support added for Email campaigns via unsubscribe link (token)
    Backup management interface added
    Interface created for AMD configurations
    Interface created for system logs
    Campaign results to csv export support added

    How Voice Broadcasting works

    User upload a list of telephone numbers, upload audio message or record his voice message through telephone , configure outbound voice gateways and start a new campaign according to requirements using ICTBroadcast web interface and within seconds, ICT Broadcast starts broadcasting user's voice message to given list of telephone numbers with real time statistics.

    WHMCS integration with ICTBroadcast For Autodialer Billing

    ICTBroadcast has released a new billing module for ICTBroadcast Service Provider edition. This module will allow WHMCS to be used as Client Management and Billing front-end for Auto Dialer service, After integration with WHMCS system will be able to provide a complete business platform for broadcasting services including Website, package listing, automated order and Account provisioning. Following are few billing scenarios which can be achieved by ICTBroadcast for voice, fax, sms and email broadcasting business.

    ICTBroadcast Deployment Scenarios

    Automated Telemarketing
    Enterprise grade message Broadcasting
    Emergency notification system
    Interactive voice broadcasting / Smart Predictive dialer
    Customer surveys / Collections
    Polling Auto Dialer
    Mass Communications / notifications
    Political voice broadcast
    Robocall / call blasting,
    Phone reminders
    Community / Emergency alerts
    School Notifications
    Non-profit Fund Raising
    Wedding invitations
    cold calling
    mass broadcasting.
    Appointment reminders
    Retail sales / Business advertisement

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