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  • 07/13/16--08:19: Old News
  • This page lists all the old VoIP news stories from the home page.


    June 2016


    May 2016


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  • 07/13/16--13:25: Bicom Systems
  • Image

    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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  • 07/13/16--22:24: voip-info.org
  • Welcome to the VOIP Wiki - a reference guide to all things VOIP.


    This Wiki covers everything related to VOIP, software, hardware, VoIP service providers, reviews, configurations, standards, tips and tricks and everything else related to voice over IP networks, IP telephony and Internet Telephony.

    Your contributions are welcome, please read the How to add information to this wiki page and the Posting Guidelinesbefore you post.


    NEWS


    News Resources


    Getting Started


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  • 07/13/16--22:43: ThirdLane
  • Image

    05/05/2016 - New version 7.5.2.1 of Thirdlane Business PBX, Thirdlane Multi Tenant PBX, and Thirdlane Elastic Cloud PBX has been released

    Third Lane Technologies, established in 2003 and located in Silicon Valley, provides unified communications solutions to government and public organizations, businesses of any size, Internet Telephony Service Providers, and Call Center Operators. Reliability, advanced features, open architecture and great value made Thirdlane products the clear choice for thousands of customers and partners worldwide.

    Recently released Thirdlane end user applications are free and available for download. Thirdlane Mobile Dialer can be installed from Google Play and Thirdlane Web Dialer Chrome Extension from Chrome Web Store.

    More information about Thirdlane Applications is available on Thirdlane Applications page on Thirdlane web site.


    Thirdlane Products


    Thirdlane offers professional unified communications software solutions for hosted and on-premises deployment. Thirdlane software solutions include Thirdlane Elastic Cloud PBX (for large scale hosted deployment), Thirdlane Multi Tenant PBX for smaller ITSPs, Thirdlane Business PBX for businesses and Internet Telephony Service Providers (for on-premises or dedicated hosted deployment), as well as software for Contact Centers.

    Thirdlane software solutions are built with standard, proven open-source components, including Kamailio SIP Server, Asterisk® PBX platform and CentOS® Linux. Thirdlane provides the best of both worlds: the freedom to choose your own devices and customize your system to fit your needs, plus the quality, security, and reliability that comes from professionally-designed, managed, and supported software.


    Try Thirdlane for Free!


    Try a demo of Thirdlane Multi Tenant or Business PBX, or download a free trial of either PBX product from the Thirdlane website.





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  • 07/14/16--16:46: 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


    CLI commands

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

    Return codes

    Always returns 0.

    See also



    Asterisk | Configuration | The Dialplan - extensions. ...

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  • 07/16/16--18:21: call.center softphone
  • Welcome to call.center™
    call.center™ is specifically designed to be the only phone app that you will ever need. This means that we incorporate all the key components for robust business, professional and personal communications into one seamless app, with plentiful features integrating standard and advanced telephony services.

    The call.center™ app presents a unique and innovative drag-to-call user interface (UI), where all the necessary functions are displayed front and center, smoothly integrating and optimizing voice operations with the workflow processes.

    call.center™ is available for multiple operating system platforms including iOS, Android and Windows, allowing you to standardize on this app wherever you go.

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  • 07/18/16--00:05: REVE Systems
  • Image

    REVE Systems ensures the best returns on technology investments and strengthens the service providers' market presence by providing them with best-in-class VoIP solutions. We have a large pool of engineers who are experienced and well trained on varied environments and cross vendor platforms. This enables us to provide 24x7 Platinum Level Support to our clients and to ensure that their services are always available to their end customers.

    Headquartered in Singapore, REVE has its major development center in Bangladesh and India, branch offices in HongKong, USA and United Kingdom. We currently service customers in over 78 countries, where more than 2600 VoIP and telecommunication service providers have placed their trust on us.

    History


    REVE Systems started in 2003 with a focused approach to serve the IP based communication industry. A Telecommunication and Software Solution provider, REVE Systems has a wide assortment of products, ranging from backbone infrastructure to peripheral products, including middleware. The company today holds a leadership position in Mobile VoIP, Softswitch& Billing and Bandwidth Optimization solutions.

    Products



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    Cisco SPA525G2 5-Line Business IP Phone with Enhanced Connectivity and Media for a New Level of User Experience



    MKI04001.jpg


    Highlights


    • Full-featured and stylish business IP phone supporting up to two Cisco® SPA500S Expansion Modules (32 button attendant consoles)
    • Cisco Mobile Link: Bluetooth enhanced integration with mobile phones to make and receive calls, import your personal contacts, and charge your mobile phone
    • Enhanced network connectivity with Power over Ethernet (PoE), 802.11g Wi-Fi client with Wi-Fi Protected Setup (WPS), and Bluetooth headset support
    • Graphics-rich, high-resolution 3.2-inch QVGA 320 x 240 color screen
    • Cisco AnyConnect VPN Client: Highly secure Internet phone connection for remote users that is simple and easy to set up
    • MonitorView for monitoring up to four video surveillance cameras from your phone
    • Cisco XML services framework: Support for productivity applications directly on your phone
    • Support for multimedia functions, such as playing MP3s, displaying digital photos, and viewing RSS feeds
    • Wideband audio for unsurpassed voice clarity and enhanced speaker quality
    • Support for both Session Initiation Protocol (SIP) and Smart Phone Control Protocol (SPCP) with the Cisco Unified Communications 500 Series for Small Business



    Overview


    The Cisco SPA525G2 5-Line IP Phone with Color Display is an excellent choice for businesses that require an enhanced user experience with a hosted IP telephony service, an IP private branch exchange (PBX), or a large-scale IP Centrex deployment. Part of the Cisco Small Business Series, the SPA525G2 uses industry-leading SPA voice over IP (VoIP) technology from Cisco, with high-quality hardware providing additional connectivity via Bluetooth, PoE (802.3af), or a Wireless-G client (802.11g).

    Standard Cisco SPA525G2 features include five active lines, VLAN-capable dual switched Ethernet ports, 802.3af PoE support, a 3.2-inch QVGA color display, a full-duplex, high-quality speakerphone, a Bluetooth interface, a Wireless-G (802.11g) client, a 2.5-mm stereo headset port, and a USB 2.0 host port. Each line can be configured independently to use a unique phone number (or extension) or can use a shared number that is assigned to multiple phones. The power supply for the SPA525G2 is sold separately. The optional SPA500S 32-button attendant console adds up to 64 buttons for receptionist positions.

    The Cisco SPA525G2 IP Phone further improves the user experience with VPN and video surveillance applications. It includes an embedded AnyConnect Secure Sockets Layer (SSL) VPN client that allows remote users to securely connect to their phone system and make calls over the Internet, without the need for additional hardware. The SPA525G2 also provides users the ability to view video feeds from Cisco WVC2300 and PVC2300 Business Internet Video Cameras, allowing users to quickly see different locations around the business in order to improve physical security. The SPA525G2 is part of the Cisco SPA500 Series IP Phones, a robust portfolio of small business phones providing a rich user experience that includes HD voice, on-phone applications, and intuitive menu options.

    The Cisco SPA525G2 provides encrypted signaling, media, and provisioning information, using state-of-the-art technologies such as Session Initiation Protocol (SIP) over Transport Layer Security (TLS), Secure Real Time Protocol (SRTP), and HTTPS to secure communications between the phone and service provider. Cisco SPA Secure Remote Provisioning provides a highly secure mechanism for the service provider to remotely manage the phone/user configuration and software upgrades.

    The Cisco SPA525G2 IP Phone can also be used with productivity-enhancing features such as VoiceView Express and Cisco XML applications when connected to the Cisco Unified Communication 500 Series in SPCP mode. ...

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  • 07/19/16--21:29: How to start a VOIP Business
  • The first thing to do is decide what part of VOIP marketplace you want to serve. Here are some possibilities:

    • VOIP Provider services
    • VOIP consulting
    • Independent Sales/Service Agent for existing VOIP service providers
    • Value Added services with VoIP
    • etc.

    Some general suggestions:

    • Pick an area that plays to your strengths. For example, if your strength is sales and marketing, pick an area where you can leverage those abilities
    • Learn all you can about the maketplace
    • Attend industry tradeshows
    • Read industry magazines, blogs, forums, etc
    • Read books
    • Do market research - talk to your potential customers
    • Ask questions
    • Test the waters — to the extent possible try before you buy, test the waters before making large commitments of time or money

    Value Added services

    If you have experience with VoIP or already in VoIP business, you can get benifit / new customers by introducesing some value added services on VoIP. Few value added services are mentioned in following, Within each service there are many choices.
    • PBX sales and service
      • Hosted PBX
      • Virtual Numbers
      • Hosted IVR / Auto attendents
      • etc

    • Message broadcasting / Call Center Solutions

    • Prepaid Cards
      • Retail prepaid cards from existing wholesale providers
      • Start your own brand of prepaid cards using services from existing wholesale providers
      • Start a new prepaid card provider company
      • Create new software package for prepaid card services
      • Create a Free Phone Booth
      • etc.


    Asterisk SIP Trunking - US — Wholesale Reseller User Portal and Admin Portal. With this service you can customize your own Admin and User portal to sell SIP Trunk Services to your end users. This means that you can set your own rates to your customers, your admin portal will charge your customers in real-time so you never have to chase your money. All services are turned up in real-time for your end users. This includes DID ordering, SIP Trunks, vFAX and all other telecom related services. This is the ultimate reseller portal program. As for the requirements you must have an Authorize.net Account and subscribe to the CIM for remote PCI protection of credit card numbers. When your customers pay the money is sent to your bank for each nightly batch settlement. The funds are transferred directly to you and our services are billed to your account at a wholesale rate. Call now: 877-686-4787 or visit us:

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  • 07/26/16--21:44: 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. (The SIM card contains all the information needed to identify network subscribers. ...

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  • 07/27/16--09:39: 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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  • 07/29/16--12:17: Hanlong Technology
  • HTek (Hanlong Technology Co. Ltd.)


    Htek Logo.jpg




    HTek Profile


    HTek™ is a name brand of Hanlong Technology Co., Ltd., a world-class designer and manufacturer of enterprise IP phones and VoIP products. HTek IP phones deliver superb sound quality, a rich set of SIP telephony features, and broad interoperability with leading phone system providers, including Broadsoft®, 3CX®, Elastix®, FreePBX®, Asterisk®, Bicom®, and Alcatel-Lucent®. All HTek IP phone products feature the Texas Instruments® (TI) chipset for crystal-clear HD sound, and are backed by an industry-leading two-year warranty. HTek is the new standard for quality and value in IP phones.

    Sold in over 50 countries worldwide, HTek products offer high-quality, cost-effective solutions that can also be easily rebranded or customized to meet OEM or ODM requirements. Since 2005, Hanlong Technology has provided enterprises, OEMs, and ITSPs worldwide with millions of advanced VoIP products. The latest HTek UC900 series IP phones continue the tradition of focusing on world-class quality, cutting-edge features, and competitive pricing.

    HTek R&D Team


    HTek has an experienced in-house R&D and engineering team to create the highest quality product designs, easily respond to OEM and ODM requests, and provide superior customer technical support. With an average of over 8 year's experience in the communication field, the R&D team's expertise and professionalism guarantee superior products and support. HTek is also proud to partner with

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    Htek Logo-med.jpg


    HTEK UC862 EXECUTIVE HD COLOR GIGABIT IP PHONE


    860.png


    Physical Features


      • 3.5”TFT-LCD, 480 x 320 pixel, 262K colors
      • Ethernet: Dual Gigabit Ethernet ports
      • Keys: 47 keys including 14 programmable keys
      • Full-duplex speakerphone with AEC
      • Handset: 1 RJ9 (4P4C)
      • Headset: 1 RJ9 (4P4C)
      • Power adapter (included): 5V/1.2A
      • Power over Ethernet(PoE): IEEE 802.3af

    VoIP Protocols

      • SIPv2, SDP(RFC 2327), RTP(RFC 1889,1890), RTCP
      • RFC 2833 X-NSE Tone Events for SIP/RTP, AVT Tone
      • Events for SIP/RTP

    Voice Codecs

      • HD wideband codec: G.722
      • HD Codec, HD speaker, HD handset
      • G.711u-law/a-law, G.723.1, G.726, G.729A/B.
      • DTMF(In-Band, RFC2833, SIP Info)
      • Acoustic Echo Cancelation(AEC)
      • Acoustic Gain Control(AGC)
      • Voice Activity Detection(VAD), Comfort Noise insertion

    Security

      • HTTPS Server/Client
      • Transport Layer Security (TLS)
      • SRTP (RFC3711), SIPS
      • VLAN QoS (802. ...

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    842.png


    Physical Feature


      • 3.5”TFT-LCD, 480 x 320 pixel, 262K colors
      • Keys: 37 keys including 4 programmable keys
      • Handset: 1 RJ9(4P4C)
      • Headset: 1 RJ9(4P4C)
      • Power adapter: 5V/1.2A
      • Power over Ethernet(PoE)

    VoIP Protocol

      • SIPv2, SDP(RFC 2327), RTP(RFC 1889,1890), RTCP
      • RFC 2833 X-NSE Tone Events for SIP/RTP, AVT Tone
      • Events for SIP/RTP

    Voice Codecs

      • HD wideband codec:
      • G.722. HD Codec, HD speaker, HD handset
      • Full-duplex speakerphone with AEC
      • G.711u/a-law, G.723.1, G.726, G.729A/B.
      • DTMF(In-Band, RFC2833, SIP Info)
      • Acoustic Echo Cancelation(AEC),
      • Acoustic Gain Control(AGC)
      • Voice Activity Detection(VAD), Comfort Noise insertion

    Security

      • HTTPS Server/Client
      • Transport Layer Security (TLS)
      • SRTP (RFC3711), SIPS
      • VLAN QoS (802.1pq)

    Telephone Interfaces

      • 4 VoIP Accounts
      • Menu-driven user interface, XML Idle Screen, Theme,
      • Screen Sleep
      • Call hold, Call waiting, Call forward, Call return,
      • Redial,Call transfer
      • Caller ID display, DND, Auto-answer, 5-Way Conference
      • Mute, Speed dial, SMS, Voicemail, Message Waiting
      • Indication (MWI) LED, Call history
      • BLF/BLA
      • Tone scheme, Volume control
      • Ring tone selection/Import/Delete
      • Broad and Deep Interoperability
      • Soft keys programmable
      • Phonebook, Black list XML/LDAP phonebook

    Network Protocol

      • TCP, UDP, ICMP, RARP, ARP, DNS, NTP,SNTP, STUN, UPnP,SNMP
      • Static/DHCP/PPPoE
      • TFTP/DHCP/PPPoE client
      • Telnet/HTTP/HTTPS server

    QoS

      • Layer 2(802.1Q, 802. ...

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  • 07/29/16--13:31: Hanlong HD IP Phone-UC804P
  • 804.png


    Physical Feature


      • Dual color LCD, 128 x 96 pixel
      • Ethernet: Dual-port Gigabit Ethernet ports
      • Keys: 37 keys including 4 programmable keys
      • Handset: 1 RJ9(4P4C)
      • Headset: 1 RJ9(4P4C)
      • Power adapter: 5V/1.2A
      • Power over Ethernet(PoE)

    VoIP Protocol

      • SIPv2, SDP(RFC 2327), RTP(RFC 1889,1890), RTCP
      • RFC 2833 X-NSE Tone Events for SIP/RTP, AVT Tone
      • Events for SIP/RTP

    Voice Codecs

      • HD wideband codec:
      • G.722. HD Codec, HD speaker, HD handset
      • Full-duplex speakerphone with AEC
      • G.711u/a-law, G.723.1, G.726, G.729A/B.
      • DTMF(In-Band, RFC2833, SIP Info)
      • Acoustic Echo Cancelation(AEC),
      • Acoustic Gain Control(AGC)
      • Voice Activity Detection(VAD), Comfort Noise insertion

    Security

      • HTTPS Server/Client
      • Transport Layer Security (TLS)
      • SRTP (RFC3711), SIPS
      • VLAN QoS (802.1pq)

    Telephone Interfaces

      • 3 VoIP Accounts
      • Menu-driven user interface, XML Idle Screen, Theme,
      • Screen Sleep
      • Call hold, Call waiting, Call forward, Call return,
      • Redial,Call transfer
      • Caller ID display, DND, Auto-answer, 5-Way Conference
      • Mute, Speed dial, SMS, Voicemail, Message Waiting
      • Indication (MWI) LED, Call history
      • BLF/BLA
      • Tone /Volume control
      • Ring tone selection/Import/Delete
      • Broad and Deep Interoperability
      • Soft keys programmable
      • Phonebook, Black list XML/LDAP phonebook

    Network Protocol

      • TCP, UDP, ICMP, RARP, ARP, DNS, NTP,SNTP, STUN, UPnP,SNMP
      • Static/DHCP/PPPoE
      • TFTP/DHCP/PPPoE client
      • Telnet/HTTP/HTTPS server

    QoS

      • Layer 2(802.1Q, 802. ...

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  • 07/29/16--15:53: Vonage Business UK Reviews

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  • 08/01/16--02:18: 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.
    • Autom8 Enterprise Automated predictive dialer from Autom8 Group that is fully Omni Channel, communicate with your customers using Email, SMS, Webchat and Social Media (Facebook, Twitter, Linkedin, Google) making this truly a solution for the future.
    • 3CX Phone System Pro offers CRM integration, advanced reporting and queue strategies, inbuilt failover, integrated video conferencing and much more. ...

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  • 08/01/16--05:22: 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


    Vox Switch
    Vox Switch is carrier grade softswitch platform from Voxvalley Technologies that is soon becoming a favorite choice for service providers and carriers across the globe. Vox Switch facilitates service providers to carry out routing, billing, reporting & monitoring of their VoIP services.
    Vox Switch platform is flexible to allow hassle free migration from other softswitches without any data loss.
    Get a Vox Switch Free Quote

    Vox Switch includes numerous advanced features
    ● Multi-level reseller user management.
    ● Multi-branding support.
    ● DID Management.
    ● Tariff plan, Balance, Payment and Voucher management.
    ● Advanced Routing.
    ● Invoicing & Billing.
    ● Real-time Reporting.
    ● IVR Routing.
    Get a Vox Switch Free Quote


    Image

    iTel Switch
    iTel Switch is a single Softswitch platform for global Retail, Wholesale, Calling card & Call shop business. Being a highly customizable and scalable VoIP Softswitch with integrated billing, it serves as an ideal platform for all the VoIP service providers who want to provide a wide range of VoIP services.

    Get a iTel Switch Free Demo:

    Key Features

    ● Rate Plan Management
    ● Flexible & Advanced Routing
    ● Live Call Monitoring
    ● Multilayer Security
    ● Analysis & Reporting
    ● Behavioral Based Alarm System
    ● Call Simulator for Route Checking
    ● Client Management
    ● DID Management
    ● Payment Management




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  • 08/02/16--20:40: REVE Systems
  • Image

    REVE Systems ensures the best returns on technology investments and strengthens the service providers' market presence by providing them with best-in-class VoIP solutions. We have a large pool of engineers who are experienced and well trained on varied environments and cross vendor platforms. This enables us to provide 24x7 Platinum Level Support to our clients and to ensure that their services are always available to their end customers.

    Headquartered in Singapore, REVE has its major development center in Bangladesh and India, branch offices in HongKong, USA and United Kingdom. We currently service customers in over 78 countries, where more than 2600 VoIP and telecommunication service providers have placed their trust on us.

    History


    REVE Systems started in 2003 with a focused approach to serve the IP based communication industry. A Telecommunication and Software Solution provider, REVE Systems has a wide assortment of products, ranging from backbone infrastructure to peripheral products, including middleware. The company today holds a leadership position in Mobile VoIP, Softswitch& Billing and Bandwidth Optimization solutions.



    REVE Systems Blog Posts






    Products


    • ALL-In-One Solutions

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  • 08/02/16--21:59: VoIP Hardware
  • This page lists information about VoIP hardware and VoIP hardware products. For phones and hardware to use with Asterisk, including VoIP phones (both hard and soft phones) and Analog Telephone Adapters, see Asterisk phones.

    PSTN Interface cards (analog, GSM, ISDN-PRI and R2/MFC)


    This section contains VoIP hardware for connecting analog or digital phone lines from the Public Switched Telephone Network to your Asterisk server. Please keep VoIP hardware providers in alphabetical order.

    .e4 VoIP Hardware


    2-Day Direct

    • Cisco SPA303 3-line business-class IP phone; Connects directly to an Internet telephone service provider or to an IP private branch exchange (PBX)

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