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  • 08/01/17--05:42: Bicom Systems


  • Bicom Systems provides the Communicating World with the most Complete Turnkey Communication Systems available by Creating, Unifying and Supporting the Most Advanced of Current Technologies.

    History


    Bicom Systems Ltd. was founded in 2005 to exploit its PBXware product.
    PBXware was the first Commercial Turnkey Telephony System to use Open Source software including Asterisk.

    Among the first customers to use PBXware was Redhat. The business model of Bicom Systems does hold similarities to Redhat in the manner by which it wraps Open Source software in a professional and charge for model, warranted to work.

    In 2008 Bicom Systems delivered a custom built conferencing solution to NASA to facilitate the holding of scientific study groups such as the Inter Planetary Conference.

    In 2009 Bicom Systems launched its Multi-Tenant Edition of PBXware.

    In 2010 Bicom Systems began a relationship with NEC to provide a hosted Telephony Platform to businesses across Australia.

    Bicom Systems published How to Grow an ITSP.


    Products



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  • 08/02/17--13:42: Asterisk Cmd Voximal
  • Voximal()

    Synopsis


    Execute a VoiceXML document over Asterisk (Based on the Voximal VoiceXML browser).
    The application use Asterisk internal API (Prompt / DTMF / Record) and installed applications.
    It replaces the old Vxml application.
    Voximal have been developped by Ulex Innovative Systems

    Description

    Voximal(url|account reference)

    Features :

    - Audio (play and record gsm, wav, WAV files)
    - Video (play and record h263, mp4, 3gp files)
    - DTMF (bargein support)
    - Transfer (use Dial/Transfer applications and to exchange with Asterisk function/variables too)
    - Text To Speech (most TTS supported with HTTP connector, and Festival/Flite and unimrcp applications, Cloud TTS (voxygen, cereproc, microsoft...) )
    - Automatic Speech Recognition (Nuance, Lunenvox, Verbio, Vtech, VoiceInteraction, Vestec, use Asterisk Speech API or unimrcp )
    - Speech To Text (Google Voice, Bing microsoft...)
    - Accounts for hosting (ranges, url, stats, max limitations)

    Documentations

    - Web site
    - Installation guide
    - Developer guide

    Configuration files

    - voximal.conf

    After execution, the VoiceXML result passed with the <exit> tag and the property ‘expr’ is accessible with the variable VOXIMAL_RESULT, the value can be configured to set the CDR userfield too.

    Asterisk Dialplan example

    [incoming]
    exten => s,1,Answer
    exten => s,n,Wait(1)
    exten => s,n,Voximal(http://download.voximal.com/examples/index.vxml)
    exten => s,n,Hangup

    VoiceXML syntax

    Download VoiceXML examples from voximal wiki or from github :


    CLI commands

    - voximal show version
    - voximal show license
    - voximal show configuration
    - voximal show statistics

    Return codes

    Always returns 0.

    See also


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  • 08/02/17--13:56: i6net
  • Image

    VoiceXML for everybusiness | www.i6net.com

    I6NET provides a VoiceXML browser for Asterisk : Voximal (formerly VXI*) :

    Products


    VoiceXML Browser for Asterisk Voximal


    The Voximal VoiceXML browser for Asterisk® gives operators and solution providers the ability to rapidly develop and deploy innovative voice and video applications via IP, PSTN, and 3G-324M networks. Voximal is fully compliant with the W3C's VoiceXML 2.0+ specification and is integrated with automatic speech recognition (ASR) and text-to-speech (TTS) software to enable advanced voice and video solutions, and real-time video calling applications. Voximal can be installed in common hardware configurations, providing a highly scalable base system to meet all customers' business and technical VoiceXML requirements.

    Voximal VoiceXML interpreter works directly with the Asterisk® PBX software supported by Digium®. Not only can users of the open source PBX run VoiceXML applications in the same server, they can now offer these powerful, scalable IVR / IVVR solutions at an affordable cost.

    Written in C, Linux Operating System, License Commercial

    Key features and benefits

    • VoiceXML V2.0/V2.1 compliant
    • CCXML replaced by Call Control functions of Asterisk
    • Base on OpenVXI voice browser template
    • Application for Asterisk PBX (app_vxml)
    • VoiceXML accounts managment configuration (for hosting services)
    • Support plugging objects with the VoiceXML tag, SDK with API available
    • Fax support (send and receive)
    • Video silence parameter with VoiceXML syntax
    • Text-to-Speech (TTS) connector included
    • Automatic-Speech-Recognition (ASR) connector included
    • Text-to-Video (TTV) with HTTP connector (option)
    • Voice-Silent Detection (VSD)
    • MRCP v1 and v2 thru uniMRCP for TTS and ASR.

    Cloud Ready:

    • We offer Amazon, Azure and Docker virtual images
    • Amazon Polly and Lexa
    • Azure Cognitive Services (TTS and STT)
    • Google TTS and STT Apis
    • IBM Watson

    Text-to-Speech engines supported:

    • Flite - free (TTS) from CMU Speech Group
    • Voxygen (TTS)
    • Cepstral (TTS)
    • Verbio (TTS)
    • Nuance Scansoft (TTS)
    • Acapela (TTS)
    • Universal HTTP connector for any TTS engines
    • Universal MRCP connector for any TTS engines (uniMRCP)

    Automatic-Speech-Recognition engines supported:

    • Lumenvox Speech Engine (ASR)
    • Verbio Speech Server (ASR)
    • VoiceInteraction Speech Engine (ASR)
    • Vestec Speech Engine (ASR)
    • Loquendo Speech (ASR-MRCP)
    • Nuance Speech (ASR-MRCP)
    • PocketSphinx (SpeechToText)
    • universal MRCP v1 and v2 (ASR) (uniMRCP modified with extra features)

    Asterisk requirements:

    • Asterisk 1.8 / 11 / 13 / 14

    Distributions: Debian, CentOS, Ubuntu, Raspbian



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  • 08/02/17--23:58: TeleOss
  • SMS Gateway Provider - TeleOSS

    TeleOSS Messaging Suite is a converged delivery platform for all messaging channels viz. SMS, USSD, MMS, IVR, eMail and eFax. Messaging suite is ready to integrate platform with standard protocols, modules and central control management interface to support all business and technology needs of service provider. Messaging Suite offers messaging gateway, messaging router, messaging hub, messaging controller, consent gateway, two-way communication platform.

    The platform has been designed based on service oriented approach where any new service and business can be implemented quickly and new revenue streams are enabled. TeleOSS enterprise grade platform is robust and scalable to integrate with complete stack of communication protocols for SMS, MMS, USSD, IVR and e-mail over HTTP, HTTPs, XML, SMTP, SMPP, SIP, SS7, SIGTRAN, SSL, RTP interfaces.

    TeleOSS Messaging Gateway

    A leading messaging gateway, TeleOSS SMS Gateway Software enables fast and reliable message delivery for service providers/aggregators over standard messaging channels for A2P and P2P traffic. A cost effective and easy to use interface for all operations, routing rules, live traffic monitoring dashboard with exhaustive traffic analysis. Bundled with rich features convergent platform for multiple messaging channels SMS, MMS, USSD, IVR, e-mail and eFax available for deployment as per service provider needs. Explore the gateway variants below.

    TeleOSS Messaging Hub

    Platform offers integrated interconnection model for messaging, enabling service & network providers (MNO/MVNOs) to reach globally. The platform is build on robust architecture with resilience and scalable to cater ever demanding need of messaging infrastructure. Dynamic rule based routing enables minimum queuing time and no message loss with deployment options of multiple instances, geo-redundant and exhaustive traffic monitoring & analysis are key advantages of platform.

    TeleOSS Messaging Router

    A carrier grade platform, supporting high availability requirements, with modular and intelligent rule based engine. The platform provides flexibility to configure routes for message delivery based on as per various parameters and logical combination of the same. Routing considers rule based approach, load balancing and SMSC capacity, source and destination of messaging.

    TeleOSS Messaging Controller

    The platform enabling MNOs/MVNOs to handle bursts in A2P and P2P traffic, with scalable and reliable architecture. The platform with rich features with flexibility of rule based service & routing mechanism provides operators the desired quality of service. On-board live traffic monitoring, alarming and reporting capabilities with quick action interface makes it easy to use. The platform supports any combination of mobile networks.

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  • 08/03/17--15:05: Thirdlane Multi-Tenant PBX


  • Thirdlane Multi Tenant PBX is a reliable, feature rich, scaleable and cost-effective 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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    Business VoIP Providers - Compare and Choose a Business VoIP Provider

    Quality business VoIP providers today offer a wide variety of feature packages, services and prices. Selecting the ideal provider and service options will depend on your type and size of business, features needed and projected volume of usage. These metrics determine if your better off paying per minute, per user, per line, or per feature. Before choosing your VoIP provider, it is essential to first determine your company's precise telecommunications needs. A current phone bill or CDR record is a great place to find the information you need to compare providers. the basics your will need are: Quantity of phone numbers, toll free numbers, user extensions, lines, calling usage for local, long distance and Toll free calls as well as and special feature to enable timely and cost-efficient initiation of your service. By consulting your chosen Voice over IP service team and seeking their expert advice in advance, you can be prepared to take the following steps to facilitate the smooth, productive startup of your services:

    • Evaluate Your Internet Connection. - Determine the strength and capacity of your Internet connection and bandwidth. You need to ensure that your system has adequate speed to best accommodate your new VoIP installation for top quality service.
    • Assess Your Company Budget and Needs. - With knowledge of your company's current budget and VoIP needs, you can more easily select the service provider and feature options that meet your requirements.
    • Determine Your Equipment Needs. - Evaluate your current and near future VoIP equipment needs. Phones can be purchased from around $50 to $500 or more. Once you decide which feature options are immediate requirements and which ones can be added later as needed, you are ready to choose your service provider.
    • Compare VoIP Providers. - By comparing VoIP company service options, advanced features and equipment along with user and industry reviews, you can best make a wise decision, selecting the ideal VoIP provider for your enterprise.

    Important Information to Request from Any Potential VoIP Provider

    Before signing a service contract with any business VoIP provider, be sure to request basic service information and practices in writing. You need to be certain of such details as startup costs and monthly fees, any limitations and costs on portable phone numbers and exactly which features are included in the service package you select. You also need to know if international calling is included, charges for adding extra features and the extent of customer care and technical services provided. Also important are such issues as whether your provider offers a money back guarantee and if there are any cancellation fees. It is also helpful to determine prior to signing up for VoIP services if there are any hidden fees assessed by your chosen provider.

    Take Full Control and Advantage of Your VoIP System

    Once your new business VoIP system and service are in place, you and your staff members will have full-control capabilities for use of your business communications system. Your service provider will ensure connection with your online portal for customizing your telecom options. These modern digital portals are user-friendly, enabling feature changes and additions to be made for immediate availability. You and your staff can make decisions and changes in real-time that work for you right in the moment.

    You can manage your call settings remotely, directing calls to voicemail or having them transferred to another number or extension. You can also make exceptions to any chosen setting in your phone system. For example, if you are expecting an important business call and want to take that call, but hold all other calls for a few hours, you can set your phone to direct only the designated call to ring on your extension. This system allows and encourages you to take complete control of your telecommunications systems and settings so that the service works for your best interests and immediate needs at all times.

    Major Business Benefits and Advantages of Installing VoIP

    With an excellent quality VoIP system installed and running well in your company offices to provide remote access for you and your employees, you can work much more efficiently, achieving more in less time. You will enjoy the many benefits of knowing that you can leave the responsibility of your advanced office telecommunications system operations to your VoIP provider while you handle other important business matters. ...

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  • 08/03/17--23:57: Old News
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  • 08/04/17--20:25: Thirdlane Business PBX


  • 08/03/2017 - New version of Thirdlane Connect for unified communications 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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  • 08/07/17--14:20: 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.

    Overview

    customLogo.gif.png

    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.

    Architecture

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


    Design

    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

    Support

    Chrome WebRTC Development Team

    Discussion List: https://groups.google.com/group/discuss-webrtc
    Google Plus Page: https://plus.google.com/113817074606039822053
    Chrome WebRTC Issue Tracker: http://code.google. ...

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  • 08/07/17--14:27: WebRTC vs VoIP
  • Theres is a bit of confusion in the telecommunication industry as to whether or not WebRTC is compatible with or runs against VoIP. WebRTC is a viable Internet Protocol (IP) communications system that parallels and runs alongside the internet-based phone system VoIP. VoIP services and WebRTC solutions are both heavily promoted in the business and residential fields. So the confusion lays here: are VoIP and WebRTC providers friends or foes? Can the two systems coexist, do they overlap, and how does the client benefit from these?


    The similarities

    WebRTC and V.VoIP are similar in that both aim to enhance the user experience and enable any consumer device (whether it be mobile phone, fax, internet etc.) to effortlessly connect from anywhere and on any network internationally.

    The differences

    The primary difference between the two services is that VoIP uses a multitude of variants such as VoIP over DSL/cable modem, voice over Wi-Fi/3G (VoWiFi/3G), voice over LTE (VoLTE), and Rich Communication Suite (RCS), while WebRTC is solely focused on browser-based communications.

    VoIP

    VoIP is an online telecommunications system which offers simpler and more efficient technology than traditional phone service. VoIP uses advance phone technology in order to make phone calls from the office or home more cost effective and with more features. Standard telephone systems uses telephone lines to transmit phone calls, using physical circuits for connection. Since VoIP is cloud-based, calls are sent as digital data and no cables are needed to send the call so any kind of Internet connection can be used to make calls and from a plethora of devices. Millions of people and businesses have switched to VoIP in order to save money as well as to be able to access the same lines from any place and any device.

    Benefits that most VoIP providers include are: around the clock customer service; reduced costs compared to traditional phones; no installation or service fees; free ad-ons including unlimited calling to the US and Canada, unlimited extensions, 1,000 free toll-free minutes, high-definition video-conferencing, desktop integration with popular CRMs, online PBX controls, virtual extensions, remote access, auto-attendants, and unlimited extensions for multiple office locations.

    WebRTC

    WebRTC (Web Real-Time Communication) is an API being drafted by the World Wide Web Consortium (W3C). Put simply, its a software intermediary that makes it possible for application programs to interact with each other and share data. WebRTC is used to enable browser-to-browser applications for voice calling, P2P file sharing, and video chat without plugins. WebRTC is an emerging technology that are accessed with JavaScript APIs and currently in development are an audio and video data stream as well as API which allow for two or more users to communicate browser-to-browser, real-time gaming, text chat, file transfer and other online based sharing.

    Connecting vs. clashing

    WebRTC makes it feasible for web developers to enable VoIP into their Web-based applications. Since WebRTC is in its early stages of development, it does not include any signaling protocol which leaves this choice and development and integration to the developer. By integrating a signaling protocol into WebRTC, a developer can create a full VoIP soft client on a browser.
    One nice example of such VoIP soft client is CryptoVoIP SIP WebRTCDialer which uses SIP Protocol for signalling. The good part of CryptoVoIP web dialer is that it does not require web-sockets or webrtc support in SIP Servers or softswitch. ...

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  • 08/10/17--01:55: Hosted VoIP Softswitch
  • Softswitch in VoIP industry is the device that connects IP to IP calls. Softswitch also acts as a VoIP Server. This provides all the call related functions.

    There are 2 types of Softswitch.

    • Class 4 softswitch
      • Class 5 softswitch.

    One of the most important function of the Softswitch is the billing software. This provides incomparable accuracy and saves the tedious process of billing manually. Apart from billing, a Softswitch also helps in routing, reporting and monitoring in VoIP business. Softswitches are also customizable according to the user necessity. Being a software completely, softswitches are less costly and does not require maintenance charges.
    One has to be careful while choosing a softswitch. An established stable company will provide better quality service than the newly found ones. Softswitch are popular for their compatibility and flexibility. So user friendly aspect is an important point to keep in mind while choosing a softswitch.

    There are 2 types of ownership for a Softswitch. One, the complete ownership. Here the owner buys the softswitch and gains total authority over it. And, the other is the monthly rental basis. In the monthly rental basis, one has to pay a pre-fixed amount every month by the given last date to continue the services of the softswitch.
    Global VoIP Service Provider, INAANI, provides iTel Switch Plus for its clients. It is available in monthly rental basis. The Switch is present in the package along with a mobile dialer. INAANI has various packages for its Hosted VoIP Services. The packages consist of the main requirements for a VoIP business like,

    1. iTel Switch Plus
    2. Byte Saver tunnelling software
    3. TP Smart Dialer or iTel mobile dialer express
    4. PC dialer

    This package is customizable and helps in getting started with Hosted VoIP business.

    Get the hosting package HERE.



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  • 08/10/17--20:53: 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


    • DIDNumberHub.com 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

    Angola

    • 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 freehttps://www.multitel.net/pbx|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.

    Argentina

    • 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.
    • BuyDDINumbers.com Provides Cheapest Argentina DID /Virtual Phone Numbers/DDI Numbers @_€ 2.95/month including free PBX. with the Free forwarding to Skype ,Gtalk , Trixbox ,Asterisk or VOIP. Phone Numbers all over the world are available. Free PBX __. Toll Free Number Available without Monthly commitments.

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  • 08/11/17--02:29: Mobile VoIP
  • Mobile VoIP is an efficient, low-cost way to communicate using your cell phone and the services provided by your home or business VoIP provider.

    How Does Mobile VoIP Work?


    Mobile VoIP works with a cell phone’s 3G, 4G, GSM, or other Internet service to send voice calls as digital signals over the Internet using voice over IP technology. Mobile VoIP phones can also take advantage of WiFi hotspots to eliminate the calling costs of a cellular voice or data plan.

    By using VoIP, mobile VoIP phone users — especially smartphone users — can benefit from lower costs when calling, texting, or other common smartphone activities. Digital data transmission using VoIP is also typically faster, as the data is spread out over multiple packets, each taking the fastest route to its intended destination.

    Using a mobile VoIP phone with WiFi hotspot access can also reduce a mobile VoIP phone user's costs by sidestepping the carrier's expensive 3G service altogether. For instance, with a cellular carrier's monthly data plan, callers can easily exceed bandwidth maximums, incurring overage charges. Tapping into WiFi hotspots with mobile VoIP software reduces that risk and extends the lifespan of the monthly data allotment.

    A mobile VoIP phone service can eliminate the need for a basic voice plan, as well as optional (and costly) text add-ons. With a mobile VoIP phone, cell phone users can enjoy more flexibility in calling times than a cellular voice plan provides, with fewer restrictions. VoIP mobile phone service means that a mobile VoIP user can make unlimited inexpensive or free calls using voice over IP technology at any time.

    Mobile VoIP users don't need to worry about the limitations associated with cell phone calling plans, such as:

    • Anytime minutes
    • Night or weekend minutes
    • Rollover minutes
    • Roaming charges
    • Incoming call charges
    • Messaging limits
    • Mobile-to-mobile calling (check with your mobile VoIP provider, some do treat in-network calls differently)

    Mobile VoIP phone users can also take advantage of the additional, integrated features a mobile VoIP app supports. This includes high-bandwidth activities such as group chat and video chat. Accessing these functions without mobile VoIP software (by fring or Talkonaut, for instance), typically requires a separate app, and using it could impact or exceed monthly text and bandwidth maximums.

    Accessing Mobile VoIP

    Cell phone users can use mobile VoIP service on their phone with the addition of mobile VoIP software. These are apps offered by VoIP phone service providers customers may already be using at home or at work, such as Vonage, or standalone mobile VoIP apps such as Skype, Vyke, or Truphone.

    Some services, such as Truphone, also offer an entire mobile VoIP network by combining a SIM (Subscriber Identity Module) card and an app together. ...

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  • 08/02/17--23:58: TeleOss
  • SMS Gateway Provider - TeleOSS

    TeleOSS Messaging Suite is a converged delivery platform for all messaging channels viz. SMS, USSD, MMS, IVR, eMail and eFax. Messaging suite is ready to integrate platform with standard protocols, modules and central control management interface to support all business and technology needs of service provider. Messaging Suite offers messaging gateway, messaging router, messaging hub, messaging controller, consent gateway, two-way communication platform.

    The platform has been designed based on service oriented approach where any new service and business can be implemented quickly and new revenue streams are enabled. TeleOSS enterprise grade platform is robust and scalable to integrate with complete stack of communication protocols for SMS, MMS, USSD, IVR and e-mail over HTTP, HTTPs, XML, SMTP, SMPP, SIP, SS7, SIGTRAN, SSL, RTP interfaces.

    TeleOSS Messaging Gateway

    A leading messaging gateway, TeleOSS SMS Gateway Software enables fast and reliable message delivery for service providers/aggregators over standard messaging channels for A2P and P2P traffic. A cost effective and easy to use interface for all operations, routing rules, live traffic monitoring dashboard with exhaustive traffic analysis. Bundled with rich features convergent platform for multiple messaging channels SMS, MMS, USSD, IVR, e-mail and eFax available for deployment as per service provider needs. Explore the gateway variants below.

    TeleOSS Messaging Hub

    Platform offers integrated interconnection model for messaging, enabling service & network providers (MNO/MVNOs) to reach globally. The platform is build on robust architecture with resilience and scalable to cater ever demanding need of messaging infrastructure. Dynamic rule based routing enables minimum queuing time and no message loss with deployment options of multiple instances, geo-redundant and exhaustive traffic monitoring & analysis are key advantages of platform.

    TeleOSS Messaging Router

    A carrier grade platform, supporting high availability requirements, with modular and intelligent rule based engine. The platform provides flexibility to configure routes for message delivery based on as per various parameters and logical combination of the same. Routing considers rule based approach, load balancing and SMSC capacity, source and destination of messaging.

    TeleOSS Messaging Controller

    The platform enabling MNOs/MVNOs to handle bursts in A2P and P2P traffic, with scalable and reliable architecture. The platform with rich features with flexibility of rule based service & routing mechanism provides operators the desired quality of service. On-board live traffic monitoring, alarming and reporting capabilities with quick action interface makes it easy to use. The platform supports any combination of mobile networks.

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  • 08/15/17--01:34: Asterisk func strreplace
  • Synopsis

    Replace instances of a substring within a string with another string.

    Description

    Searches for all instances of the <find-string> in provided variable and replaces them with <replace-string>. If <replace-string> is an empty string, this will effectively delete that substring. If <max-replacements> is specified, this function will stop after performing replacements <max-replacements> times.

    NOTE: The replacement only occurs in the output. The original variable is not altered.

    NOTE: varname should be not a variable value or function return value, but a variable name or function name with params:
    • same => n,NoOp(${STRREPLACE(TRUNK,"_","")})
    • same => n,NoOp(${STRREPLACE(CDR(disposition),"","_")})

    Syntax

    STRREPLACE(varname,find-string[,replace-string[,max-replacements]])


    Arguments

    Not available

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  • 08/15/17--01:48: VoIP Termination
  • Please add information to this page about VoIP call termination.

    What is VoIP Termination?

    VoIP call termination is used to refer to the procedures that are used for routing telephone calls from one provider to the next provider until the call has been routed to the last telephone company and has been received by the recipient. Voice termination is another term that is used for call termination. The telephone companies are also referred to as providers or carriers.

    Called Party

    The called party is the person who has received the telephone call. The end point of the route may be on the Internet or may be at a point that was reached by routing the call through the public switched telephone network. The procedures for routing the call stop when the call has been received by the recipient. The process may seem simple to individuals who do not experience problems with making telephone calls but is not so simple to individuals who make calls that will not connect such as calls to a different country.

    Calling Party

    The calling party is the person who has initiated the call and who wants to be connected to the called party. There may be problems with the telephone calls that begin on the Internet and end at a cellular phone. The sound of the voices may seem to be amplified with echoes and have a scratchy quality. The calls that are initiated with Google Voice may never connect or could be delayed.

    VoIP

    Voice over Internet protocol (VoIP) is a term that is used to describe a call that was initiated on the Internet such as Skype calls or calls made with Google Voice. The calls that were initiated on the Internet usually end at a point that is not on the Internet. Most of the recipients of telephone calls receive the calls with a landline phone or with a cellular phone. The route may begin on the Internet but will end at a point that was reached by routing the call from the public switched telephone network (PSTN), which is the common description for call termination.

    Internet Networks

    A tier-one operator is licensed and registered to operate an Internet protocol (IP) network for Internet telephony services. Tier-one operators can handle call origination and call termination. A tier-two operator can lease services from a tier-one operator. The tier-three operators can lease services from either a tier-one or two-two operator. There are also resellers of VoIP services and wholesalers in the market. The quality of the services is not very high because of the inconsistencies in the market such as fluctuations with demand, fraud and problems with doing business on an international level.

    Call Origination

    Call origination is used to refer to telephone calls that originate from the public switched telephone network and end the route on the Internet. Call termination is considered to be the opposite of call origination because the direction of the paths are reversed. The terms are associated with the starting point of the calls, path of the route of the calls and termination point of the calls. The operators of IP telephony services can handle calls that originate or terminate on the Internet.

    Fees

    The fees for services are subject to the regulations of several countries because the routes of the calls will cross over more than one country. Those countries can use legislation to control the fees for the services. Termination rates are usually very high for the countries in the Middle East and Asia. The rates are intentional high because there are more incoming calls than outgoing calls from the countries, which is caused by the diaspora effect of a migrating population.

    VoIP Termination Providers

    Please list VoIP Termination providers here in alphabetical order.



    1Pipe Telecom | OnePipe is a licensed carrier providing Hosted PBX and SIP Trunking in over 60 countries. We have regional network facilities on five continents connected across our private network. ...

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  • 08/18/17--10:42: VoIPSupply.com
  • About VoIPSupply.com


    VoIPSupply.com is the retail website of North America's leading VoIP solutions provider, VoIP Supply. VoIPSupply.com offers a fast and easy way to find the VoIP hardware you are looking for.

    VoIPSupply.com has products for consumers, small medium businesses, enterprises, government/educational entities, resellers and service providers. You can visit VoIPSupply.com 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.

    VoIPSupply.com Offerings


    VoIPSupply.com 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 VoIPSupply.com
    • 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


    VoIP Supply Partner Program Webinar

    Join our Partner Program webinar to learn about our new partner portal and sign up to sell hosted VoIP through the CloudSpan MarketPlace! Check out our four training sessions:

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    Here is a list of producers of ready-made, black box PBXs that are based on Asterisk (in no particular order):


    Since the globalization has enforced many enterprises and thus, it is essential to upgrade enterprises with advanced yet modern systems. Ecosmob renders various cost-effective IT services to its clients worldwide. Its VoIP services and solutions are highly scalable and engineered on two core elements – quality and reliability.

    Key Advantages for Asterisk IP PBX Solution:

    - Easy to setup, configure and maintain
    - Offers better customer productivity and services
    - Significant cost savings using VoIP providers
    - Web/GUI based configuration
    - Scalable, reliable and efficient
    - Eliminates phone wiring and vendor lock-in
    - Wide array of features

    For Asterisk IP PBX Solution Contact: https://www.ecosmob.com/, 1-303-997-3139

    Arrowtel
    -Specializing in asterisk based solutions
    -Local installs in Southern California, USA - Support Worldwide
    -Provides design and support of any Asterisk-based PBX of all sizes
    Location: CA 90025, USA.
    Telephone:800-284-3842
    E-mail :info@arrowtel.net
    Web : http://www.arrowtel.com



    Bicom Systems


    Bicom Systems is the creator of Telco-in-a-Box, the most complete IP PBX Unified Communications solution for ITSPs, Service Providers, CLECs, and more. Telco-in-a-Box meets all of your telecom needs including telephony, billing, mobility, security, and more in one tidy package. With unmatched compatibility, stability, and reliability, Telco-in-a-Box is the tool to build and grow a telecom. The product suite includes:

    • SERVERware is a Cloud IP Services Delivery Platform
    • PBXware is an IP-PBX turnkey business communications platform
    • TELCOware is a provisioning platform that performs billing, accounts, and more
    • sipPROT is an advanced SIP security module
    • sipMON is a SIP monitoring module that analyzes and improves VoIP calls
    • gloCOM is a desktop and mobile Unified Communications app

    A Zycoo Asterisk 13 System



    A Zycoo Asterisk V.13 base VoIP phone systems is user friendly and license free IP PBX system, includes optional telephony interfaces: PSTN, GSM, WCDMA, E1/T1, ISDN BRI. ...

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  • 08/20/17--22:33: IP PBX
  • IP PBX is a phone system that utilizes IP communications. Traditionally IP PBX's are located on site where they can also interface to traditional telco services such as analogue phone lines. The business end users connect via IP to the IP PBX for voice service.


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  • 08/21/17--06:02: Call Center Software
  • Call center software is the software system that allows a company or organization to run a call center. This page lets you compare call center software providers.

    There are hundreds of different providers of call center software across the globe, and every call center software system has its pros and cons. When selecting the right call center software for your business, contact center, or call center, it's important to decide which features you want your phone system to have.

    Types of Call Center Software


    ACD helps productivity by assigning inbound agents to incoming calls. The automatic call distributor uses a set of instructions to determine who gets the call in the system. The algorithm can route calls based on agent skill or whoever has an idle phone. ACD can use caller ID or automatic number identification, but usually interactive voice response is enough to help the system determine the reason for the call.

    An automatic call distributor can also take advantage of computer telephony integration. Agents can receive relevant data on their computers along with the incoming call.

    Computer telephony integration is a broad category of software that connects telephone and computer systems. Computer telephony integration software can have both desktop and server functions. Various applications make up a system that can control phones, display call information, and route and report calls.

    Interactive voice response allows callers to route themselves to the appropriate department or use the company’s database for assistance. More sophisticated interactive voice response systems can access accounts and perform certain tasks, such as activating a credit card through a bank’s phone system. IVR involves using dial tone multi-frequency or voice commands. In the VoIP industry, a PBXauto attendant is near interchangeable with IVR. However, auto attendants are not capable of speech recognition.

    A predictive dialer calls a list of phone numbers at once. Outbound agents are then connected to the numbers that answer. A predictive dialer uses calculations to minimize the idle time of agents and the potential of losing answered calls when no agents are available.

    Contact Center Software

    For contact centers, software includes applications for chat, email, and web interaction in addition to telephony functions.

    Call Center Software Providers

    This is a list of call center software providers and developers. Please keep this list in alphabetical order.

    • Dialer360 is Call Center Software which provides you inbound, outbound and blended services on high voltage and low cost for your call center. It is the complete telecom solution including Omni Channel, communication with your customers by using Email, SMS, Web chat and Social Media (Facebook, Twitter, LinkedIn, Google) etc. It can be integrated with the CRM of your choice. It gives you services for 24/7 and make sure your data save. It will make your call center to compete with the modern technology in the market. ...

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