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  • 02/04/16--05:01: VOIP Service Providers B2B
  • Here is a list of VOIP Service Providers focusing on Business-To-Business services. This includes VoIP origination and VoIP termination, plans aimed at call centers, IVR providers and generic Asterisk users. See also:


    Services which require the use of locked ATA devices should not be listed on this page. Nor should services which do not permit simultaneous calls — most services here support at least 4 simultaneous incoming calls. Please list only services which support Asterisk connections, via SIP or IAX2, to the PSTN.


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  • 02/04/16--17:39: IP Office for SOHO
  • UC510 new.jpg

    ZYCOO Releases New UC Series IP Office for SOHO


    Chengdu, China (December 10, 2015) --- ZYCOO, the leading developer and manufacturer of SMB IP Phone Systems, is proud to announce the release of the updated version of UC Series IP Office for SOHO.

    This release has successfully passed lengthy and vigorous testing procedures and comprises of improvements to functionality, product stability and interface design. This latest UC Series is the smartest and most compact device we have released to date, and is a much more appropriate solution for small and home office(SOHO) environments requiring combined IP PBX and WiFi router functionality.

    Aimed directly at SOHO market, certain enterprise functionality such as call queue, peer trunk mode, auto answer, DISA, Smart DID, call back, VLAN, IPv6 have been deleted as these are not required for this target market.
    1-E.jpg


    However, we have added some new functionality including G.729 audio codec which guarantee’s clear communication but at reduced bandwidth and optimization of wireless functionality to guarantee the stability of wireless connection.

    “As simple as possible” is the purpose of the newest UC series to provide a perfect solution to simplify usage, configuration and maintenance for SOHO customers.
    SOHO Solution2.jpg


    Highlights:

      • 10 Extension Users
      • 3 Concurrent Calls
      • High Definition Voice
      • Batch-Add Extensions
      • DialPlan/ IVR/ IVR Prompts
      • Call Forward/ Transfer/ Parking
      • VoiceMail (VMail to Email)
      • Friendly User Interface
      • Strengthened Security
      • VPN Server/Client(L2TP/PPTP)
      • Remote Update&Management
      • Allow Up to 8 Wi-Fi Terminals to Access
    For more information, please visit: http://zycoo.com/html/IP_Office_for_SOHO.html


    Contact us:


    ZYCOO China
    Web: www.zycoo.com
    Tel: +86 (28) 85337096
    Address: 7F, B7, Tianfu Software Park, Chengdu, China.

    ZYCOO UAE
    Web: www.zycoo.ae
    Tel: +971 (4) 3798839
    Address: #305, Al Riffa Plaza Bldg., Khalid Bin Alwaleed RD, DUBAI, UAE

    ZYCOO UK
    LS21, Armstrong House, First Avenue, Robin Hood Airport, Doncaster, UK (DN93GA)


    ALL THE RIGHTS ARE RESERVED BY ZYCOO Co.,Ltd

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  • 02/05/16--00:13: Free VoIP Networks
  • Whats a Free VoIP Network?
    Free VoIP Networks are based on the SIP.EDU project, which means:

    • It's a 100% SIP compliant network that allows you to connect with SIP softphones/IP phones/WiFi Phones...
    • It's a network where you dont pay to register and get a SIP URI.
    • Allows you to freely dial people inside your domain.
    • Allows you to freely dial people from other Free VoIP Networks.
    • Allows you to freely receive calls from other Free VoIP Networks.
    • Does ENUM lookups to Arpa, E164.org and nrenum.net before dialing E164 numbers.

    Free VoIP Networks available

    • BBee-free calls between BBee-Users, free mobile applications, Free text and pictures Messaging, free calls between iPhone and Android users.
    • MO-Call-free calls between MO-Callers, free mobile applications, Instant Messaging, free calls between iPhone users, callback calls,access numbers...
    • StarTel - StarTel.pt - does ENUM lookups, Instant Messaging, Video, Presence and free PSTN calls to 30 countries
    • VoipGate - IAX2, SIP, Enum, various option......
    • OnSIP - Free SIP to SIP calls, free extension to extension dialing. Unlimited free users and and extensions. Loads of free features including time based routing. Built on open standards means OnSIP works with any SIP compliant phones and networks. WebRTC and SIP over WebSocket support.
    • TalkBD - SIP only. Free SIP phones.Voice mail,Conference
    • Free SIP providers - Free SIP providers. Review of free sip services.
    • CloudNumbers - Free calls between CloudNumbers customers and to other users on the TeleWare platform (deposit required for outbound calls). 0844 number allocated for inbound calls. Other numbers available.
    • http://www.vpn3000.com free calls for sip or iax2 between users - Free calls between users using sip or iax2, also allows the use of sipbroker shared dids, use your own device or soft phone. no monthly fees.
    • FreelyCall provides free internet voice and video calls, in addition to low rate local and international calls to regular telephones and mobiles
    • Sipmobile Free calls to USA and Canada. Free US number. Free audio and video calls between Sipmobile users, free calls to Sipbroker networks. Instant Messaging. PSTN access numbers. Cheap international calls to regular telephones and mobiles. ENUM lookups. WebRTC support. Free WebRTC client Webphone. ...

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  • 02/05/16--06:32: Digium
  • 2009-07-16_085739.png

    Digium®, Inc., the Asterisk® company, is the original creator and primary developer of Asterisk®, the industry's first open source telephony platform. Digium provides hardware and software products, including AsteriskNOW™, the complete open source software appliance; Asterisk Business Edition™, the professional-grade version of Asterisk; and the Asterisk Appliance™, hardware-based telephony solution, to enterprises and telecommunications providers worldwide. Digium also offers a full range of professional services, including consulting, technical support, and custom software development.

    Used in combination with Digium's telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over IP, TDM, switched and Ethernet architectures. Digium's offerings include VoIP, conferencing, voicemail, legacy PBX, IVR, auto attendant, media servers and gateways, and application servers and gateways.

    History

    Mark Spencer founded Linux Support Services in 1999 while still a Computer Engineering student at Auburn University. When faced with the high cost of buying a PBX, Mark simply used his Linux PC and knowledge of C code to write his own! This was the beginning of the world-wide phenomenon known as Asterisk, the open source PBX, and caused Mark to shift his business focus from Linux support to supporting Asterisk and opening up the telecom market. Linux Support Services is now known as Digium, and is bringing open source to the telecom market while gaining a foothold in the telecom industry.

    Digium is based in Huntsville, Alabama.

    Philosophy


    Digium is a young and fast-moving company with an energetic and hip culture that reflects the philosophy of Asterisk and the open source revolution. Mark strongly believes that every technology he creates should be given back to the community. This is why Asterisk is fully open source. Today that model has allowed Asterisk to remain available free of charge, while it has become as robust as the leading and most-expensive PBXs.

    Digium was founded on the notion that the customer should have control over the technology that goes into his telecommunications systems, a rare notion in the proprietary world of Telecom. Since the inception of Asterisk, the idea of open source communications technology has been revitalizing an industry which was at one time crippled by the dominance of monolithic dinosaurs.

    Digium employees find an added joy in their work knowing that they are pioneering this revolution. The Digium offices are filled with innovative problem-solvers in every department, not just in engineering, because fresh thought has been key to Digium's success from the very beginning.

    Website: http://www.digium.com
    Store: Digium Store
    Sales: http://www.digium.com/en/forms/contact_sales.php
    Support: http://www.digium.com/support
    Telephone (North America and Worldwide): +1 256 428 6000
    Telephone (EMEA): +44 207 183 7577
    Facsimile: +1 256 864 0464
    Toll Free (North America): 1 877 344 4861 / 1 877 DIGIUM1
    IAXTel: 700 428-6000
    Skype: digium

    Job Openings:


    Work@Digium

    Products:


    Complete List

    Hardware


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  • 02/05/16--06:34: Yealink SIP-T18P
  • Yealink SIP-T18P Simple, Affordable and Easy-to-use IP Phone



    yealink-sip-t18p750x750.jpg


    The SIP-T18P is the simply IP phone that equipped with TI TITAN chipset,comprehensive telephony features, PoE, auto provision, and interoperable
    with the leading IP-PBX and soft switch. It is economical and can be desktop or wall mounted.

    Designed for working environment needing a basic feature IP phone, the SIP-T18P is a very cost-effective choice for small business, SOHO and Hotel,
    college apartment, supermarket, and warehouse etc.


    Highlights


    • TI TITAN chipset and TI voice engine
    • 2 programmable keys, Auto provision
    • 1xLAN, 1xdual-color LED, Phone label
    • Power over Ethernet, Wall-mountable


    Phone Features


    • 1 VoIP account, Hotline
    • Call waiting, Call transfer, Call forward
    • Call hold, Mute, Redial, DND
    • 3-way conferencing, Speed dial
    • Direct IP call without SIP proxy
    • Volume control, Ringtone selection
    • Tone scheme, System log export
    • Integrated Voice Response System


    IP PBX System Integration


    • Music on hold
    • Call park, Call pickup
    • Dial plan, Dial now
    • Voicemail
    • Message Waiting Indication (MWI)
    • Distinctive ringtone


    Voice Features


    • Wideband codec: G.722
    • Narrowband codec: G.711u/A, G.726, G.729AB, G.723.1
    • VAD, CNG, PLC, AJB, AGC


    Network Features


    • SIP v1 (RFC2543), v2 (RFC3261)
    • NAT Traversal: STUN mode
    • DTMF: In-band, out-of band (RFC2833) and SIP INFO
    • Proxy mode and peer-to-peer SIP link mode
    • IP Assignment: Static/DHCP
    • TFTP/DHCP client
    • Telnet/HTTP server
    • DNS client


    Management


    • Built-in HTTP web server
    • Configuration: browser/phone/auto-provision/IVR
    • Auto provision via TFTP/FTP/HTTP/PnP
    • Auto provision for firmware, configuration, ringtone etc.


    Security


    • QoS: IEEE 802.1p/q tagging (VLAN), Layer 3ToS
    • Digest authentication using MD5/MD5-sess
    • Secure configuration file via AES encryption
    • Admin/user configuration mode


    t18p.jpg

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  • 02/05/16--06:35: Yealink SIP-T20P
  • Yealink SIP-T20P Cost-effective, Enterprise 2-line IP Phone



    yealink-sip-t20p.jpg


    Yealink expands its lineup of IP phones with a new entry-level product SIP-T20P. It is equipped with TI TITAN
    chipset and 2x15 characters LCD, offers 2 VoIP accounts, high-definition voice, broad range of voice codecs,
    security protection for privacy, rich features including XML phonebook, headset, PoE, PnP Auto-provision, and
    seamlessly work with the leading IP-PBX and soft switch.

    It allows users to make calls in a simple, convenient and reliable manner and fully meet the requirement in
    which the basic business features are required. What is more, SIP-T20P is easy to install and inexpensive to
    start up for corporate office and residential users.



    Highlights


    • TI TITAN chipset and TI voice engine
    • 2 VoIP accounts, 2x15 characters LCD
    • HD Voice: HD Codec, HD speaker, HD handset
    • IPV6, BLF/BLA
    • 2xLAN, PoE, Headset, Wall-mountable


    Phone Features


    • 2 VoIP accounts, Hotline, Emergency call
    • Call hold, Call waiting, Call forward, Call return
    • Call transfer (blind/semi-attended/attended)
    • Caller ID display, Redial, Mute, DND
    • Auto-answer, 3-way conferencing
    • Speed dial, Voicemail
    • Message Waiting Indication (MWI) LED
    • Tone scheme, Volume control
    • Direct IP call without SIP proxy
    • Ring tone selection/import/delete
    • Phonebook (300 entries), Black list
    • Call history: dialed/received/missed/forwarded
    • Menu-driven user interface
    • Localized language and input method
    • Soft keys programmable


    IP PBX System Integration


    • Busy lamp field (BLF), BLF list
    • Bridged line appearance (BLA)
    • DND&Forward synchronization
    • Intercom, Paging, Music on hold
    • Call park, Call pickup
    • Call recording, Call completion
    • Group listening, Group pickup
    • Anonymous call, Anonymous call rejection
    • Network conference
    • Distinctive ringtone
    • Dial Plan, Dial-now


    Codecs and Voice Features


    • Wideband codec: G.722
    • Narrowband codec: G.711µ/A, G.723.1, G.726, G. ...

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  • 02/05/16--06:36: Yealink SIP-T22P
  • Yealink SIP-T22P Professional IP phone with 3 Lines & HD voice



    yealink-sip-t22p.jpg



    Yealink SIP-T22P features intuitive user interface and enhanced functionality which make it easy for
    people to interact and maximize productivity. With TI TITAN chipset and TI leading VoIP voice engine,
    it enables enhanced high-definition audio, outsourced management options, flexible deployment and
    third-party communications applications. As a cost effective IP solution, it helps users to streamline
    business processes, delivery a powerful, security and consistent communication experience for small
    and large offices environment.

    Highlights


    • TI TITAN chipset and TI voice engine
    • 3 VoIP accounts, 132x64 graphic LCD
    • HD Voice: HD Codec, HD speaker, HD handset
    • BLF, XML Browser, Hot-desking
    • 2xLAN, PoE, Headset, Wall-mountable

    Phone Features


    • 3 VoIP accounts, Hotline, Emergency call
    • Call hold, Call waiting, Call forward, Call return
    • Call transfer (blind/semi-attended/attended)
    • Caller ID display, Redial, Mute, DND
    • Auto-answer, 3-way conferencing
    • Speed dial, SMS, Voicemail
    • Message Waiting Indication (MWI) LED
    • Tone scheme, Volume control
    • Direct IP call without SIP proxy
    • Ring tone selection/import/delete
    • Phonebook (300 entries), Black list
    • Call history: dialed/received/missed/forwarded
    • Menu-driven user interface
    • Localized language and input method
    • Soft keys programmable


    Advanced Features


    • XML phonebook search/import
    • LDAP phonebook
    • XML Browser


    IP PBX System Integration


    • Busy lamp field (BLF), BLF list
    • Bridged line appearance (BLA)
    • DND&Forward synchronization
    • Intercom, Paging, Music on hold
    • Call park, Call pickup
    • Call recording, Call completion
    • Group listening, Group pickup
    • Anonymous call, Anonymous call rejection
    • Network conference
    • Distinctive ringtone
    • Dial Plan, Dial-now


    Codecs and Voice Features


    • Wideband codec: G.722
    • Narrowband codec: G.711µ/A, G.723.1, G.726, G. ...

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  • 02/05/16--06:37: What is VOIP
  • Introduction

    VOIP is an acronym for Voice Over Internet Protocol, or in more common terms phone service over the Internet.
    If you have a reasonable quality Internet connection you can get phone service delivered through your Internet connection instead of from your local phone company.

    Some people use VOIP in addition to their traditional phone service, since VOIP service providers usually offer lower rates than traditional phone companies, but sometimes doesn't offer 911 service, phone directory listings, 411 service, or other common phone services. While many VoIP providers offer these services, consistent industry-wide means of offering these are still developing.

    How does VOIP work?

    A way is required to turn analog phone signals into digital signals that can be sent over the Internet.
    This function can either be included into the phone itself (See: VOIP Phones) or in a separate box like an ATA .

    VOIP Using an ATA


    Ordinary Phone ---- ATA ---- Ethernet ---- Router ---- Internet ---- VOIP Service Provider

    VOIP using an IP Phone


    IP Phone ----- Ethernet ----- Router ---- Internet ---- VOIP Service Provider

    VOIP connecting directly

    It is also possible to bypass a VOIP Service Provider and directly connect to another VOIP user. However, if the VOIP devices are behind NAT routers, there may be problems with this approach.

    IP Phone ----- Ethernet ----- Router ---- Internet ---- Router ---- Ethernet ---- IP Phone


    Applications using VOIP

    Traditional telephony applications, such as outbound call center applications and inbound IVR applications, normally can be run on VOIP.

    Why use VOIP?

    There are two major reasons to use VOIP
    • Lower Cost
    • Increased functionality

    Lower Cost

    In general phone service via VOIP costs less than equivalent service from traditional sources. This is largely a function of traditional phone services either being monopolies or government entities. There are also some cost savings due to using a single network to carry voice and data. This is especially true when users have existing under-utilized network capacity that they can use for VOIP without any additional costs.

    In the most extreme case, users see VOIP phone calls (even international) as FREE. While there is a cost for their Internet service, using VOIP over this service may not involve any extra charges, so the users view the calls as free. There are a number of services that have sprung up to facilitate this type of "free" VOIP call. Examples are: Free World Dialup and Skype for a more complete list see: VOIP Service Providers

    Increased Functionality

    VOIP makes easy some things that are difficult to impossible with traditional phone networks.
    • Incoming phone calls are automatically routed to your VOIP phone where ever you plug it into the network. Take your VOIP phone with you on a trip, and anywhere you connect it to the Internet, you can receive your incoming calls.
    • Call center agents using VOIP phones can easily work from anywhere with a good Internet connection. ...

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  • 02/05/16--06:38: Yealink
  • yealink_sidebar.png


    Yealink IP Phones are designed for the user who wants a great quality phone at a great price. Business users will love the new Yealink IP Phones’ large LCD display, superb HD audio quality, and simple user interface; administrators will love the intuitive configuration process and cool features like built-in packet capturing utility, and no re-boot configuration changes; resellers will love the generous volume discounts and great margin (call us!).

    Yealink SIP-T20P Phone Features

    Yealink-SIP-T20(P)-Picture-01.jpg

    • 2 VoIP accounts, hotline, emergency call
    • Call waiting, call transfer, call forward
    • Hold, mute, flash, auto-answer, redial
    • 3-way conference, DND, speed dial
    • XML phonebook import/export, call history
    • Volume adjustment, ring tone selection
    • Tone scheme, System log
    • Multi-language (more than 20)

    Yealink SIP-T22P Phone Features

    t22.png

    • 3 VoIP accounts, hotline, emergency call
    • Call waiting, call transfer, call forward
    • Hold, mute, flash, auto-answer, redial
    • 3-way conference, DND, speed dial
    • Phonebook (each record with 3 phone numbers, 300 entries), black list
    • XML Phonebook search/import/export
    • Lists of missed, received, dialed and forwarded calls (100 entries in all)
    • Volume adjustment, ring tone selection
    • Tone scheme, System log
    • Multi-language (more than 20)s

    Yealink SIP-T26P Phone Features

    t26.png

    • 3 VoIP accounts, hotline, emergency call
    • Call waiting, call transfer, call forward
    • Hold, mute, flash, auto-answer, redial
    • 3-way conference, DND, speed dial
    • XML Phonebook search/import/export
    • Black list, call history (100 entries)
    • Volume adjustment, ring tone selection
    • Tone scheme, System log
    • Multi-language (more than 20)
    • Supports up to 6 expansion modules

    Yealink SIP-T28P Phone Features

    t28.png

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  • 02/05/16--06:39: Cisco SPA303 IP Phone
  • Cisco SPA303 1-Line IP Phone



    data_sheet_c78-601648-2.jpg


    Highlights


    • 3-line business-class IP phone
    • Connects directly to an Internet telephone service provider or to an IP private branch exchange (PBX)
    • Dual switched Ethernet ports, speakerphone, caller ID, call hold, conferencing, and more
    • Easy installation and highly secure remote provisioning, as well as menu-based and web-based configuration
    • Supports both Session Initiation Protocol (SIP) and Smart Phone Control Protocol (SPCP) with the Cisco® Unified Communications 500 Series




    Comprehensive Interoperability and SIP-Based Feature Set


    Based on SIP, the Cisco SPA 303 3-Line IP Phone with 2-Port Switch has been tested to help ensure comprehensive interoperability with equipment from voice over IP (VoIP) infrastructure leaders, enabling service providers to quickly roll out competitive, feature-rich services to their customers.

    With hundreds of features and configurable service parameters, the Cisco SPA 303 addresses the requirements of traditional business users while building on the advantages of IP telephony. Features such as easy station moves and shared line appearances (across local and geographically dispersed locations) are just some of the many advantages of the SPA 303.

    The Cisco SPA 303 IP phone can also be used with productivity-enhancing features such as VoiceView Express, and Cisco XML applications when interfacing with the Cisco Unified Communications 500 Series in SPCP mode.



    Carrier-Grade Security, Provisioning, and Management


    The Cisco SPA 303 uses standard encryption protocols to perform highly secure remote provisioning and unobtrusive in-service software upgrades. Remote provisioning tools include detailed performance measurement and troubleshooting features, enabling network providers to deliver high-quality support to their subscribers. Remote provisioning also saves service providers the time and expense of managing, preloading, and reconfiguring customer premises equipment.


    1017240128.jpg

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  • 02/05/16--08:12: VOIP PBX and Servers
  • Please list information about VoIP PBX and Servers on this page. Please keep VoIP PBX and server provider information in alphabetical order, and below any other relevant information.

    Page Contents

    Numeric

    • 1comms.co.uk: Asterisk-based converged telephone system for UK Businesses
    • 1GATE VoIP PBX by Wangate: Cheap VoIP PBX with hardware DSP. Optional internal gateways for Analog/ISDN/PRI/GSM. VoIP resellers welcome.
    • 2daydirect: Brand NEW Small Business VoIP phones. Free 2 day shipping anywhere in the United States
    • 2N NETSTAR PBX, virtual PBX: VoIP PBX system
    • 2N Omega IP PBX: VoIP PBX system
    • 2N VoiceBlue Enterprise: Simple VoIP SIP PBX
    • 3CX: Windows IP PBX / VOIP Phone system
    • 4PSA VoipNow: Hosted PBX software for service providers and enterprises, accelerating SaaS deployment. It runs on Linux environments (RHEL, SuSE Linux, CentOS, Fedora) on x86 and Power PC architecture based servers.
    • 8ix Zenith: 8ix Zenith spells an Asterisk derived IP Telephony application with the most advanced calling and communication features.

    A

    • ALLO PSTN-IPPBX for SOHO with 30 IP extension, upto 6 Analog Extension & upto 4 PSTN trunk
    • ActivePBX™ | Turn-Key Business Phone System $149/mo.

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  • 02/06/16--15:00: 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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    514 DID's www.514dids.com (Montréal)


    Aheeva Technology inc.

    • Web site: http://www.aheeva.com/
    • E-mail:info@aheeva.com
    • Phone: (514) 223-2581
    • Products: Inbound, SoftPhone, Predictive dialer, Full digital recording

    Andre Courchesne - Consultant

    • Web site: http://www.net-forces.com/
    • E-mail:courchea@net-forces.com
    • Phone: (514) 430-7583
    • Products: Asterisk consulting, installation, custom programming, CTI integration, Broadcast dialer, Call Center dialer.

    Atelka Contact Center Solutions

    • Web site: http://www.atelka.com/
    • E-mail:info@atelka.com
    • Phone: (514) 448-4905
    • Products: Virtual Contact Center

    BGM Informatique

    • Web site: http://www.bgm.qc.ca
    • E-mail: dgagnonATbgm.qc.ca
    • Phone: (418) 668-0744
    • Products: Installation Asterisk par des professionels DCAP
    • Saguenay, Lac-St-Jean

    Canadian Voip Supplier

    • Web site: http://canadianvoipsupplier.com
    • E-mail: info AT Canadianvoipsupplier DOT com
    • Phone: +1 (855) 514-VOIP (8647)
    • Positron Telecom system experts. Consultation, sales, installation, and troubleshooting of existing installs.
    • Affordable (almost wholesale) pricing on all our voip products
    • Everything is in stock and ships from Canada. Next day delivery for most products.
    We have over 15 years experience in Montreal installing and servicing phone systems and cabling. Call us anytime! (On parle aussi Francais!)

    Communications Télésignal Inc


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  • 02/08/16--03:24: Android VOIP
  • Below you will find news and a list of VoIP apps for Android that allow you to make cheap or free VoIP calls using your data connection instead of your cellular network.

    News


    FAQ


    VOIP Native Applications for Android


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  • 02/08/16--03:25: 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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  • 02/08/16--03:25: Apple iPhone VOIP
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  • 02/08/16--05:11: Digium
  • 2009-07-16_085739.png

    Digium®, Inc., the Asterisk® company, is the original creator and primary developer of Asterisk®, the industry's first open source telephony platform. Digium provides hardware and software products, including AsteriskNOW™, the complete open source software appliance; Asterisk Business Edition™, the professional-grade version of Asterisk; and the Asterisk Appliance™, hardware-based telephony solution, to enterprises and telecommunications providers worldwide. Digium also offers a full range of professional services, including consulting, technical support, and custom software development.

    Used in combination with Digium's telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over IP, TDM, switched and Ethernet architectures. Digium's offerings include VoIP, conferencing, voicemail, legacy PBX, IVR, auto attendant, media servers and gateways, and application servers and gateways.

    History

    Mark Spencer founded Linux Support Services in 1999 while still a Computer Engineering student at Auburn University. When faced with the high cost of buying a PBX, Mark simply used his Linux PC and knowledge of C code to write his own! This was the beginning of the world-wide phenomenon known as Asterisk, the open source PBX, and caused Mark to shift his business focus from Linux support to supporting Asterisk and opening up the telecom market. Linux Support Services is now known as Digium, and is bringing open source to the telecom market while gaining a foothold in the telecom industry.

    Digium is based in Huntsville, Alabama.

    Philosophy


    Digium is a young and fast-moving company with an energetic and hip culture that reflects the philosophy of Asterisk and the open source revolution. Mark strongly believes that every technology he creates should be given back to the community. This is why Asterisk is fully open source. Today that model has allowed Asterisk to remain available free of charge, while it has become as robust as the leading and most-expensive PBXs.

    Digium was founded on the notion that the customer should have control over the technology that goes into his telecommunications systems, a rare notion in the proprietary world of Telecom. Since the inception of Asterisk, the idea of open source communications technology has been revitalizing an industry which was at one time crippled by the dominance of monolithic dinosaurs.

    Digium employees find an added joy in their work knowing that they are pioneering this revolution. The Digium offices are filled with innovative problem-solvers in every department, not just in engineering, because fresh thought has been key to Digium's success from the very beginning.

    Website: http://www.digium.com
    Store: Digium Store
    Sales: http://www.digium.com/en/forms/contact_sales.php
    Support: http://www.digium.com/support
    Telephone (North America and Worldwide): +1 256 428 6000
    Telephone (EMEA): +44 207 183 7577
    Facsimile: +1 256 864 0464
    Toll Free (North America): 1 877 344 4861 / 1 877 DIGIUM1
    IAXTel: 700 428-6000
    Skype: digium

    Job Openings:


    Work@Digium

    Products:


    Complete List

    Hardware


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  • 02/08/16--07:44: SIP Trunk Providers UK
  • This page is a list of SIP trunk providers in the UK (United Kingdom) including England and Scotland. Please keep this list in alphabetical order. UK SIP providers looking to add their services can do so in the list below.
    __

    • ALTOTELECOMCall Center VoIP provider - AltoTelecom is VoIP company that provides SIP Trunking services for Call Centers, hotels, small and large business using a PBX or an Asterisk based predictive dialer such as Vicidial or Goautodial for telemarketing sales, wholesale rates to USA, Canada and UK www.altotelecom.com CHAT Support Available

    1comms VoIP provider for UK Businesses. Sip trunk services, UK DDI's and Asterisk Support for business users and resellers. White label & fully itemised per second billing.

    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.
    • Actual licensed operator
    • Branded customer portals
    • Multiple geographic locations on one Hosted PBX
    • Coverage in over 60 countries
    • Unlimited inbound on each channel
    • Great for inbound call centers
    • Call Center and Dialer options available

    2 - Tel2 - Feature-rich VoIP provider with FREE signup and UK DIDs. Offer a host of services including call recording, web and video conferencing and collaboration services, faxmail (+ T.38 passthru), locate me, Smartphone and Desktop Apps and more. We offer all UK landline and tollfree numbers. Suitable for all users from residential, business through to large call centres. Wholesale and reseller programs and a white label 'Telco in the Cloud' product available. Lowest Rates. Save with calling bundle rates of 0.6p for landlines, 2p for Mobiles. 40+ countries at 1p/min. Build your own Telco in the cloud under your own branding and set your own rates and create your own calling plans and bundles using our fully automated web portals.

    Aloha Connect Part of the Aloha Telecommunications Group, a UK National operator. Aloha Connect provides a simple platform to provision a prepaid Free SIP trunk with options to purchase DIDs (numbers) from over 50 countries. Aloha focuses purely on the quality side of the market (especially in regards to international calling). UK and Mobile calls are some of the cheapest rates on the market.

    ALTOTELECOMVoIP provider for business and Call Centers - AltoTelecom is VoIP company that provides SIP Trunking services for Call Centers, hotels, small and large business using a PBX or an Asterisk based predictive dialer such as Vicidial or Goautodial for telemarketing sales, rates under 1 cent per minute to USA, Canada and UK www.altotelecom.com

    • AVOXI is a Sip Trunk Provider. AVOXI virtual call center solutions provide virtual call center products like SIP trunking and VoIP gateway solutions, with international toll-free numbers.


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  • 02/08/16--07:45: VoIP Providers UK
  • This page is a list of VoIP providers in the UK (United Kingdom) including England and Scotland. Please keep this list in alphabetical order. UK VoIP service providers looking to add their services can do so in the list below.

    1comms.co.uk VoIP provider for UK Businesses. Sip trunk services, UK DDI's and Asterisk Support for business users and resellers. White Label fully itemised per second billing.

    Zaplee - Buy Toll Free or any UK Local Number and forwarding to your Skype, GoogleTalk, VoIP, SIP, landline, or mobile phone. Get Enquiry on Zaplee's UK Support Team : +44 (0)20 3734-2162 .

    1VOC - 1VOC.com provides Voice over Internet Protocol (VoIP) phone service to residential, mobile and business customers worldwide.Using free software on your computer or mobile device, Analog Telephone Adapters (ATA’s), IP Phones, or IP PBX’s, along with a broadband internet connection and allows you to bypass the traditional local, long distance, and international telephone carriers resulting in significant savings for both incoming and outgoing calls. 1VOC customers receive the benefits of high quality phone calls worldwide, at the lowest rates (billing per second).

    2 - Tel2 - Feature-rich VoIP provider with FREE signup and UK DIDs. Asterisk friendly ITSP with support for the IAX2 protocol. Offer a host of services including call recording, web and video conferencing and collaboration services, faxmail (+ T.38 passthru), locate me, Smartphone and Desktop Apps and more. We offer all UK landline and tollfree numbers. Suitable for all users from residential, business through to large call centres. Wholesale and reseller programs and a white label 'Telco in the Cloud' product available. Lowest Rates. Save with calling bundle rates of 0.6p for landlines, 2p for Mobiles. 40+ countries at 1p/min. Build your own Telco in the cloud under your own branding and set your own rates and create your own calling plans and bundles using our fully automated web portals.

    Andrews & Arnold Ltd provides SIP trunks, single and multiple extensions as well as SIP2SIM where a mobile phone SIM card is provided that works as a SIP endpoint that registers against your own (or their own) SIP service.

    Advancefone provides, Business Trunks | Multi Channel DID | Fax to Email | Cheap International Phone calls , Advancefone offers great low call rates to world, call using your mobile, landline phone or pc, save up to 60% on phone bills, Receive faxes in your email with our fax enabled DIDs, no extra charge or page limit for receiving faxes. please visit: Advance Phone International for more information.

    Aloha Connect Part of the Aloha Telecommunications Group, a UK National operator. Aloha Connect provides a simple platform to provision a prepaid Free SIP trunk with options to purchase DIDs (numbers) from over 50 countries. Aloha focuses purely on the quality side of the market (especially in regards to international calling).

    ALTOTELECOM Call center VoIP Provider- AltoTelecom is VoIP company that provides VoIP services for Call Centers, hotels, small and large business ideal for telemarketing sales because of the low cost of the calls, rates are under 1 cent per minute to USA, Canada and UK

    AstraQom Business Solutions include Hosted PBX, Live Answering, Computer Telephony Integrations and Business Internet. ...

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  • 02/08/16--11:38: 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.
    • 3CX Phone System Pro offers CRM integration, advanced reporting and queue strategies, inbuilt failover, integrated video conferencing and much more.
    • Avatar Dialler Avatar Dialler is your ultimate choice for call center software. You can get complete solution of contact center dialer

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