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  • 11/14/16--04:34: Cloud PBX
  • A cloud PBX system is a based on cloud computing technology, where data is stored and transferred over the Internet, rather than on a computer or piece of hardware that an end-user owns.

    Cloud technology that has been around for many years, but has only more recently become popular for consumer use with the introduction of programs like MobileMe and the iCloud, both consumer-based cloud computing technologies.

    History of the Cloud

    Cloud technology has many applications for businesses and consumers. In the mid-1990’s, companies that had previously used “point-to-point” data circuits to move and store data began to use a cloud technology instead as they found that this was a cheaper way to do the same tasks, and made more efficient use of bandwidth.

    The term “cloud” likely refers to the drawings of clouds used by its program developers to represent the abstraction of the Internet. The symbol was used to mark the division between responsibilities of the service provider company and the responsibilities of the customer.

    Cloud technology means that you can access the data and services of your cloud from any device connected to the Internet. So, with a cloud PBX provider, you can use all of the standard PBX features without the need of an actual PBX in your home or office.

    What is PBX? What is Cloud PBX?

    PBX, or public branch exchange, is the name used to refer to the technology that any given telephone provider uses to route calls. Originally, a PBX was a large unit of hardware that had to be stored on site and operated by hand. These machines could be very costly for a company, as they were large, needed special storage facilities, needed regular repairs, and had to be operated manually.

    Companies with multiple phones and many employees can’t function properly without some kind of PBX technology in the office. In more recent years, a PBX could be made to operate automatically, without an operator, but even these devices were expensive and complicated (the cheapest are around $400). And if a business needed to move to a new location, or add or remove a phone line, the process could be very costly and time consuming.

    Initial cost and upkeep of traditional PBX vs cloud PBX

    • Traditional PBX: About $5,000 including cost of installation and new equipment fees. $10.00/hour for an operator. If managed without an operator, can still run to around $300 or $400 for upkeep and Internet connection.
    • handSIP Hosted PBX— Plans from $21.95/month include unlimited calling within US/Canada
    • RingCentral Virtual PBX: $19.99/month plus $0.049/additional minute
    • Cebod Telecom: $19.99/month All Inclusive and Unlimited Call
    • Vocalocity Virtual Extension:$14.99/month plus $0.03/additional minute
    • RingOffice Cloud Phone System: Plans from $10/month
    • Switchvox Cloud by Digium: $29/month, unlimited minutes and all UC features included
    • MultiTEL Cloud Hosted PBX— free for up to 5 users, $0.0095 for calls to US/Canada
    • VoIPstudio : A Simple and Powerful Cloud Based Business PBX System. Plans start from $ 4.99/Month
    • MySmarttel+ cloud PBX at € 7.8/month, +unlimited call recording and storage at € 6. ...

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  • 11/15/16--20:35: maple4VOIP
  • maple4voip voipinfo.jpg

    Digium OpenVox Sangoma ZYCOO VOIP solutions

    When it comes to internet telephony simply the best


    Product Catgeories

    maple4VOIP store
    Hauptstrasse 9
    83052 Bruckmühl, Germany
    Tel: (+49) 8062-726993-0
    Fax: (+49) 8062-726993-9
    Internet: www. ...

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  • 11/16/16--18:50: 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


    • Asterisk-based converged telephone system for UK Businesses
    • 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.


    • APPRIN آپرین Middle East VoIP Distributor IP Phone, IP PBX, Gateway, ATA from Digium, Grandstream, Barix.
    • 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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  • 11/17/16--07:41: VoIP Providers USA
  • This is a list of VoIP providers in the USA. These companies typically support multi-line telephone systems, small PBX gateways and hosted VoIP. Please add VoIP Providers USA to the list below.

    VoIP Providers in the USA

    NocRoom Turn-key Business Hosted PBX

    Cloud/Hosted Business PBX, NocRoomPBX will transform your communications and help you improve your productivity. We will do a full turn-key solution for you business!

    NocRoom Data Centers Miami FL, Dallas TX, Las Vegas NV and Los Angeles CA .

    • Axis-VoIP by AxisInternet AxisInternet is a Denver based residential and business VoIP phone system provider. We provide service throughout the US, Canada and Mexico. We have a state of the art hosted PBX system with user friendly, mobile friendly user portal and the best feature set found in any of today's IP PBX systems!

    • .e4 SIP/PBX/API/SMS - .e4's hosted PBX and SIP products lead the way. Check us out for a 1 month trial.

    • 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. ...

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  • 11/17/16--11:38: Call Center
  • A call center is a location with a phone system used exclusively for the purpose of receiving and making a large amount of telephone calls. A call center is designed for large volumes of calls with large numbers of workers. Frequently, call centers are designed as contact centers, allowing companies to provide documented and recorded customer service to large groups of customers.

    Large businesses use call centers to interact with their customers, including services like customer and technical support. A company can run its own call center or contract a third party for the service.

    A call center functions through operators, known as agents or sometimes customer representatives, and computerized telephony (CTI). An agent’s workstation consists of both a computer and phone, with which an agent can automatically dial numbers, transfer customers, or look at customers’ account/profile details.

    Call centers traditionally use PBX systems for call management, which can include an Automatic Call Distributor or line hunting to distribute incoming calls. A PBX may also feature voice recognition and IVR.

    Call Center Uses

    A call center can be either inbound, outbound or both.
    Outbound call centers may be used for:
    • telemarketing
    • fraud prevention
    • conducting surveys
    • debt collection

    Inbound call centers that receive calls from customers seeking assistance can be technical support or customer service.

    Businesses can use call centers for the above-mentioned customer interaction to complement their work (service, sales, billing, etc) or a business can offer products and services primarily served via interaction, such as:
    • mail-order catalogs
    • other ordering services (e.g. flower delivery)
    • telephone triage, or nurse consultation

    If a call center also handles letters, email, fax, text chat, or any web-based interaction, it is called a contact center.

    Call Center Varieties

    Call Center Systems

    To run a call center, one must first know the call volume for one’s company (if outsourcing). A smaller call center has less need of line hunting and call routing to select which agent receives a call, but it must have agents able to handle differing tasks.

    A call center needs a number of technological features. First and foremost, call centers require a call management system such as a PBX. A PBX should have Automatic Call Distribution for all call centers and Interactive Voice Response included for inbound or blended call centers. ...

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  • 11/11/16--01:39: Web Hosting
  • Please add information about web hosting and web hosting providers and companies to this page.

    Web Hosting Providers

    Please keep this list in alphabetical order

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  • 11/23/16--05:15: Yeastar - MyPBX
  • IP PBX for SMBs

    The new compact, feature rich PBX for every-day use!

    MyPBX is a standalone embedded hybrid PBX for small businesses and remote branch offices of larger organizations (2-300 users per site). MyPBX also offers a hybrid solution (a combination of VoIP applications using legacy telecom equipment) alternative for enterprises who are not yet ready to migrate to a complete VoIP solution.




    Users: 32
    Concurrent Calls(Max): 15
    Voicemail: 3000min

    Up to 4 Analog Ports (FXO/FXS)
    Up to 4 BRI Ports
    Flash: 512 MB Onboard Flash
    RAM: 256 MB Onboard RAM
    LAN: 1 (10/100Mbps)

    Size: 160x160x30 mm
    Weight: 500g
    Power Supply: AC 100~240V/50~60Hz (DC 12V, 1A)

    Protocol: SIP(RFC3261), IAX2
    Transport Protocol: UDP, TCP, TLS, SRTP
    Codec: G.711, GSM, SPEEX, G.722, G.726, ADPCM, G.729 A, H261,
    H263, H263p, H264 ,MPEG4.
    DTMF: In-band, RFC2833, SIP INFO
    LED: Red for FXO, Orange for BRI, Green for FXS.
    Network: Firewall, VLAN, DDNS, QoS, DHCP Server, Static IP, VPN, PPPoE
    Multiple Languages Support:
    Chinese, English, French, German, Hebrew, Italian, Portuguese, Russian,
    Spanish, Swedish, Turkish and more.


    Auto Provision
    Blind Transfer
    BLF Support
    CDR (Call Detail Records)
    Call Forward
    Call Parking
    Call Pickup
    Call Routing
    Call Transfer
    Call Waiting
    Caller ID
    Define Office Time
    DISA (Direct Inward System Access)
    Distinctive Ringtone
    DND (Do Not Disturb)
    Follow Me
    IVR (Interactive Voice Response)
    Intercom/Zone Intercom
    Mobility Extension
    Multi-language Prompt
    Music On Hold
    Music On Transfer
    One touch record
    PIN User ( PIN Code Control)
    Paging/Zone Paging
    Ring Group
    Route by Caller ID
    Skype Integration (Skype Connect)
    Speed Dial
    Three Way Calling
    Voicemail to Email
    Voicemail Forwarding
    Web Based Control Panel
    Spy functions (Normal Spy, Whisper Spy, Barge Spy)




    Users: 100
    Concurrent Calls: 25
    Voicemail: 3000min

    Up to 16 Analog Ports (FXO/FXS)
    Up to 8 GSM Ports(Quad-Band GSM/GPRS850/900/1800/1900MHz)
    Up to 8 UMTS Ports(UMTS 900/2100MHz or 850/2100MHZ or 850/1900MHZ)
    Up to 8 CDMA Ports
    Up to 8 BRI Ports
    Flash: 512 MB Onboard Flash
    RAM: 512 MB Onboard RAM
    USB: 1 (USB2.0)
    LAN: 1 (10/100Mbps)
    WAN: 1 (10/100Mbps)

    Size: 290x180x33 mm
    Weight: 700g
    Power Supply: AC 100~240V/50~60Hz(DC 12V, 5A)

    Protocol: SIP(RFC3261), IAX2
    Transport Protocol: UDP,TCP,TLS,SRTP
    Codec: G.711, GSM, SPEEX, G.722, G.726, ADPCM, G.729 A, H261,
    H263,H263p, H264 ,MPEG4. ...

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  • 11/24/16--21:05: 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

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  • 11/28/16--01:44: 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.


    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.


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  • 11/28/16--16:05: VoIP Providers Canada
  • This is a list of VoIP providers in Canada. These companies typically support multi-line telephone systems, small PBX gateways and hosted VoIP. Please add VoIP Providers Canada to the list below.

    Business VoIP Providers in Canada

    • 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
    • FiberConX100% Canadian owned/operated. Providing premium hosted business phone services,including: Phones, SIP Lines/trunks, Conference calling
    • REVE Systems is a Singapore based company that provides VoIP & IP Communication solutions for Mobile VoIP, SIP Softswitch, VoIP Billing, Bandwidth Optimization, WebRTC, MVNO and Mobile OTT. A RED Herring’s 2012 Top 100 Global Winner, REVE serves more than 3000 VoIP and telecommunication service providers in over 78 countries.
      • VoiceMeUp Largest Coverage for Canadian DIDs for business usage. Channel agregation available. Wholesale and Affiliates program. T.38 Supported In/Out. Supports SMS, Smart Caller ID. User accessible interface. ...

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    SecAst is a fraud and intrusion detection and prevention system designed specifically to protect Asterisk based phone systems. SecAst uses a variety of techniques and proprietary databases to detect intrusion attempts, halt ongoing attacks, and prevent future attacks. In addition, SecAst uses advanced techniques to detect valid credentials that have been disclosed / compromised and are being abused, and uses heuristic algorithms to learn attacker behaviour. Upon detection of fraud or attack SecAst can disconnect calls and/or block the attacker from Asterisk at the network level.

    SecAst offers the industry's largest databases of fraudulent phone numbers and hacker IP addresses, available to both carriers and end-users. Using Telium's comprehensive fraud & hacker databases allows administrators to mitigate the intrusion risk of known attackers, and reduce the risk of fraudulent toll / premium rate charges.

    SecAst offers a proprietary geographic IP address database allowing the administrator to create allow / deny rules (geofencing) down to the city level without large or complex firewall rules (all geofencing rules remain within SecAst). Use of geofencing dramatically reduces the number of, and risk from, attacks allowing administrators to quickly eliminate continents / countries / regions / cities where their users would never be located.

    SecAst is a 100% software solution, communicating with a number of Asterisk, network, and Linux subsystems. The data from these sources allows SecAst to monitor connection and dial attempts with invalid credentials, the rate at which users / peers are dialing, the number of channels in use by user / peer across all protocols, the source IP of remote users / peers, etc. By combining this data with Fraud and GeoIP databases SecAst can effectively stop attacks / fraud in its tracks, and alert the administrator with details of each attack.

    SecAst offers extensive interfaces to interact with other programs, utilities, external firewalls, billing systems, etc. allowing for considerable customization. For example, changes in Threat Level can trigger scripts which alert administrators, shutdown interfaces, change firewall rules, etc.

    SecAst is available in both free and commercial editions. You can get SecAst, as well as more documentation, at


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    HAAst is a software package which creates a high-availability cluster out of any pair of Asterisk servers. HAAst can detect a range of failures on one Asterisk server and automatically transfer control to the other server, resulting in a telephony environment with minimal down time.

    HAAst is a 100% software solution, with switchover in seconds. Built-in intelligent network control allows for a single IP address to be shared between peers, so clients/phones automatically connect to the active Asterisk server without change. Built-in replication and synchronization between servers also reduces maintenance and support activities.

    HAAst includes a wide range of sensors to detect failure/degredation of a peer, its hardware, the network, the connection to carriers, etc. The sensors contribute to an overall health score which allows HAAst to automatically take corrective action, notify an administrator, or worst case start an orderly transition to the standby peer.

    HAAst is an easy to use solution, with shell (command line), telnet, socket, and web interfaces, suitable for beginners and experts alike. HAAst is ideal for demanding telephony environments like call centers, emergency 911 operations, medical facilities, and mid-to-large size businesses, as well as small-businesses looking for high PBX uptime using low-cost off the shelf components.

    HAAst does not require any other High Availability/heartbeat software, nor require use of shared disk/block level device etc. In fact, HAAst does not share any hardware / logical device between peers, so there is no single point of failure. HAAst is robust in functionality yet simple to set up and use.

    HAAst is available in Free and Commercial editions. The Free edition is suitable for companies wanting to test if the basic functionality & compatibility meets their needs. In addition, the Free edition is a functional and useful high availability add-on for SOHO environments, more capable that any other DIY scripts, etc. The commercial edition is suitable for companies with critical telephony uptime requirements including call centers, 911 emergency centers, hospitals, etc.

    HAAst is available in both free and commercial editions. You can get HAAst, as well as more documentation, at

    technology_overview. ...</body> </html>

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  • 11/29/16--16:25: Asterisk RTMP channels
  • RTMP Channel driver

    This page describes chan_rtmp, an Asterisk channel driver for RTMP made by Ulex.
    The license for chan_rtmp is GPL V2.

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  • 12/06/16--02:39: SBO Bandwidth Saver
  • What is SBO?

    SBO is the short form of Synchronous Bandwidth Optimizer, it is a complete software for VoIP service provider and it's developed by Synchronous ICT, a world famous VoIP software provider. SBO mainly work to reduce the internet cost, beside this, it also has some other cool features.

    How Does SBO work?

    There are mainly two parts in SIP communication, one is payload and another is RTP header. For a SIP call with G.729 it consumes 31.5 kbps bandwidth. But, noticeable matter is that the payload size is only 8 kbps. Rest of the bandwidth is consumed by RTP and other headers. SBO has its own proprietary VoIP protocol which can replace the RTP and only transmit payload size thus reducing bandwidth consumption.

    Why Bandwidth Saver is Important for VoIP?

    Minimizing bandwidth consumption is the main purpose of SBO Solution, but, at the same time it help to reduce business operation cost. As a perfect bandwidth saver SBO cutoff more than 80% internet cost directly to protect it's clients from over paid. But, it never compromised with service quality. In VoIP telecommunication, main expenditure is Bandwidth cost. For a SIP call with G.729 codec it consumes more than 32 kbps of bandwidth. So, per 32 ports bandwidth consumption with G.729 Codec is about 936 kbps and with G.723 its about 657 kbps, while the payload size is only 8 Kbps. Rest of the bandwidth is consumed by RTP and other headers which is not important for voice call. So, a perfect bandwidth solution has capacity to reduce it without degrading the voice quality at all.
    To run a quality VoIP business, required a well maintained, congestion free, telco grade bandwidth connection. Such type of bandwidth connection is very costly as well as extremely rare. So, it was almost impossible to run quality VoIP business basically for small and medium entrepreneurs. But, now things are different, bandwidth saver with multipath facilities make it possible to run IP business with low cost share internet connection by maintaining supreme service quality. A perfect bandwidth saver should have the following key features:

    Key Features of SBO Bandwidth Saver

    Reduce Bandwidth Cost for More Than 80% Without Degrading The Service Quality;
    Increase Service Quality Dramatically;
    Increase Route Performance;
    Increase Average Call Duration by 1-2 Minutes. That means ensure good ACD and ASR;
    Work with any types of internet firewall to ensure interruption free, stable VoIP business.
    Work with all sorts of Internet Connection- 3G, 4G, WiFi, WMAX, EDGE, GPRS and so on;
    It allows using more multiple internet connection at a time to balance load among available networks;
    Work with Low Bandwidth, and manage congested network to ensure high performance.
    It is not important to arrange Static or Real IP to operate your IP business rather it work with normal/ public shared internet connections and support SIP (Session Initiation Protocol);
    SBO generally work with all Commonly use Codec, such as G.729, G. ...

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  • 12/08/16--07:56: CNAM
  • CNAM is an acronym which stands for Caller ID Name.

    When phone calls are made, there are usually two user-facing identifiable pieces of information: a phone number and a Caller ID Name (usually a 15-character string). CNAM can be used to display the calling party's name alongside the phone number, to help users easily identify a caller.

    There are numerous CNAM lookup services which allow you to pay a small fee to lookup the CNAM of a specified caller (by phone number).

    CNAM Lookup Services List:

    Sponsored:OpenCNAM - Caller ID Made Simple.

    - Get started instantly
    - No lengthy contracts
    - No drawn out sales cycle
    - Carrier grade Caller ID
    - Standard and Plus service levels
    -- Standard service offers real network CNAM at competitive rates based on volume
    -- Plus service offers nearly 90% coverage in North America including toll-free names and international coverage
    - Low latency performance
    - Integrate using: HTTP, ENUM, SS7 or SIGTRAN
    - 99.999% uptime
    - Live chat and toll free phone support 1-888-315-TELO (8356)

    Telo pioneered the first developer friendly telephony data API in 2011 with the launch of OpenCNAM. Since then, Telo has released subsequent versions of OpenCNAM that extends Caller ID coverage to over 229 countries. Today, Telo is committed to leading data services innovation by providing easy to use, enterprise level, developer friendly APIs.

    To make changes to this page, please email support. Cost: Only $0.005 per query for carriers or $0.009 for hobbyists! No catch, guaranteed with easy paypal integration. Sign Up for a FREE Account and we will credit you 30 FREE CNAM queries to try No monthly fees or account minimums and 20 free queries to test our service when you open an account ( instant setup ). Simple HTTP API or Fast AGI that can be placed in your Asterisk dial plan. Also native support for Switchvox PBX systems. Results are never cached so you get up to the minute real-time results. Retail prices are $.006 per query and bulk pricing is available with a volume commitment of at least 25,000 queries per month. Free support and installation assistance is available. offers both CNAM ($0.006) and LRN ($0.0003) look ups. No minimums and monthly charges. Simple HTTP API, easy to integrate to Asterisk dial plan.

    CID(name) Professional CNAM (Caller name) delivery

    • EVERY LOOKUP IS LIVE FROM THE SS7 (direct from the carrier owning the number)
    • NO 3rd party data sources
    • NO monthly fees
    • NEVER pay full price for unavailable results
    • Carrier grade, multi-redundant platform
    • Simple to integrate HTTP API
    • 99.7% caller id name accuracy
    • Lightning fast query responses (under 500ms)
    • Volume pricing as low as $0.002 per query
    • Try before you buy, 100 free dips with every new account
    • You choose the output, TEXT/JSON/XML
    • Track sub-accounts
    • Easy integration with Freeswitch, Asterisk, OpenSIPS, and other open source voip platforms
    • Easy access and daily downloads to your account activity
    • Thousands of happy customers

    Get CARRIER GRADE CNAM at offers both CNAM and a pseudo-CNAM service at a fraction of the cost. ...

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  • 12/09/16--02:19: VoIP Routes
  • VoIP Routes are similar to regular telecom routes, except they are designed for use with VoIP telephone systems. VoIP packets can travel over the Internet over VoIP routes, which essentially lets the VoIP call bypass the traditional telephone routes.

    Check out the table on the right side of this page to find providers of VoIP routes. You can also browse our VoIP routes forum to buy and sell VoIP routes and minutes.

    VoIP Routes Providers

    VoIP Routes testing services

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  • 12/09/16--07:24: 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.

    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

    ALMOSTFREECALLS - Almostfreecalls is one of the leading VoIP (Voice over Internet Protocol) Service Provider specialising in Internet Telephony Solutions for businesses, VoIP reseller programs and VoIP carrier services across varied areas of expertise.

    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.

    • 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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    This page is a list of SIP trunking providers in the Netherlands. Please keep this list in alphabetical order. SIP providers looking to add their services can do so in the list below.

    • 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.
    • ALMOSTFREECALLS - Almostfreecalls is one of the leading VoIP (Voice over Internet Protocol) Service Provider specialising in Internet Telephony Solutions for businesses, VoIP reseller programs and VoIP carrier services across varied areas of expertise.
    • ALTOTELECOM - 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
    • AVOXI AVOXI Virtual Call Center Solutions - VoIP Service Provider, provide virtual call center products like SIP trunking and VoIP gateway solutions, with international toll-free numbers. Contact Number 1-800-462-8694.
    • CallForwarding - Be present anywhere in the world with toll free forwarding services from Contact Number: 800-231-9802
    • CM Telecom | Call centers, IT & software firms, enterprises and wholesalers choose CM Telecom because of our carrier-class voice solution that offers excellent quality, lowest call rates, and 24/7 traffic monitoring | Contact +31 (0) 76 572 7000.
    • Dynamic Telecom - Dynamic Telecom offers affordable online phone solutions through VoIP telephone systems in the cloud.
    • - Sip solutions for everyone. Dutch, English, German and Spanish support. Virtual Phone Numbers.
    • MKB Voice - Specialist in Voice-over-IP for SMB.
    • Redworks Best Telecom operator offers high availability, unlimited SIP trunking and hosted VoIP business telephony

    Switch2Voip Voip for Call Centers and Business is a low cost wholesale VoIP leading provider for call centers small/large business looking to save on their phone calls.

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  • 12/12/16--10:52: ThirdLane
  • Image

    12/11/2016 - Thirdlane Connect and the new version of Thirdlane Multi Tenant PBX platform is available

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

    Recently released Thirdlane Connect and other Thirdlane unified communication applications are free and available for download.

    Thirdlane Connect is an advanced unified communication application with support for voice, video, private and group messaging, screen and file sharing. Thirdlane Connect is available for modern web browsers, mobile, Windows, Mac and Linux.

    Thirdlane Connect for Android devices can be installed from Google Play

    Thirdlane Mobile Dialer is a simple dialer for mobile devices. Thirdlane Mobile dialer can be installed from Google Play

    Thirdlane Web Dialer Chrome Extension integrates Thirdlane Connect with Salesforce, Zoho, Zendesk and other CRMs and is available from Chrome Web Store.

    Thirdlane Products

    Thirdlane offers professional unified communications software solutions for hosted and on-premises deployment. Thirdlane software solutions include Thirdlane Connect, Thirdlane Elastic Cloud PBX platform (for large scale hosted deployment), Thirdlane Multi Tenant PBX platform for smaller ITSPs, Thirdlane Business PBX for businesses and Internet Telephony Service Providers (for on-premises or dedicated cloud 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 platform, or download a free trial of either PBX product from the Thirdlane website.

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