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    Fail2Ban


    Fail2Ban is a standard Linux tool used to scan log files and then block IP's found in those log files using iptables. Fail2ban depends completely on the application (in this case Asterisk) to detect any intrusion/failure and log the user data, upon which fail2ban can then act. Fail2ban does not provide any type of intrusion detection, hack detection, etc., it depends completely on Asterisk to do that. As noted by Digium http://forums.asterisk.org/viewtopic.php?p=159984 fail2ban is not an intrusion detection / anti-hacking tool

    Note that as of Asterisk 13 Digium is moving towards security events through the AMI, and moving away from log files. For now fail2ban is still compatible with Asterisk but consider fail2ban a short-term solution only. See this wiki page for alternatives: Asterisk security

    You can get Fail2Ban, as well as more documentation, at www.fail2ban.org. At the time this is being written, the current release is 0.8.4.

    Fail2Ban With Asterisk


    The following describes how to setup Fail2Ban to work with Asterisk:

    SECURITY NOTE: fail2ban is rather limited in its ability to detect attacks against asterisk.
    More info http://forums.asterisk.org/viewtopic.php?p=159984
    Consider a more comprehensive product like the free edition of SecAst www.generationd.com

    Easy Install Script for Fail2ban version 0.8.4 / Red Hat


    This script was written by Cédric Brohée in order to simplify and accelerate the integration of the solution in a basic Asterisk configuration on Red Hat.
    Do not hesitate to read the bash script and make changes to match your own configuration.

    Before running it, you will have to do chmod 755. ...

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  • 05/25/15--05:55: SMS
  • SMS = Short Message Service

    SMS is a technology for passing short messages (mostly text) from one device to another. It is supported on a variety of mobile network systems, including GSM, UMTS, CDMA and iDEN. Most recently support has also started to appear on fixed networks.

    In Asterisk SMS is supported in several ways:

    • Asterisk cmd Sms is a mechanism of exchanging SMS using ETSI ES 201 912, which is a protocol for encoding SMS on landlines. This is only supported on a few selected carriers though.

    • chan_mobile can send SMS via the connected mobile phone.

    • FastSMS is an Asterisk command that exchanges SMS through a commercial gateway providing worldwide support for SMS.

    • ZIM-SMS is an Asterisk application which allows to send SMS from a dialplan as well as voicemail notifications based on information available in voicemail.conf. The application communicates with ZIM-SMS using XML-POST and is distributed under GPL as a sourcecode with detailed instructions. Please note, that you will need an account with ZIM to use their service. During the initial, test phase, only Canadian destinations will be covered.

    • jkSMS lets you send "sms" from your cell phone (or email client) to your Asterisk box. The message is read aloud via Swift (or festival) and you can reply or delete queued messages.

    See Also
    • Alaris SMS Platform— comprehensive SMS management solution comprising a 3 in 1 system for SMS trafffic switching, billing, and routing (read more...)
    • MediaCore SMS Solution - solution for handling carrier-to-carrier SMS traffic that consists of switching, routing and billing modules. It supports unlimited number of SMSs within a session.
    • CallMax retail SMS - convenient retail SMS platform within the CallMax Class 5 Solution. It enables end-users to send SMS to external mobile operators and is perfect for SMS marketing campaigns.
    • Amarrelo provides SMPP and GSM gateway Outlook like SMS client for coporate usage
    • Kannel provides an SMS & WAP gateway
    • PlaySMS: Free and Open Source SMS Gateway
    • SMSD (found in most Linux distributions)
    • FrontlineSMS is a standlone cross-platform Desktop app written in Java which can be used by NGOs on a laptop with a mobile phone to act as a messaging hub with auto-responders, group forwarding & even a J2ME surveying client.
    • SMSTools is a command-line SMS sender/receiver which can integrate with other applications...runs on Linux (or Windows via Cygwin). ...

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  • 05/25/15--06:02: BULK SMS
  • Bulk SMS, also known as bulk messaging or bulk text messaging is the act of disseminating SMS messages in large numbers so they are able to be delivered to various mobile phone terminals. This form of messaging is typically utilized by consumer brands, banks, enterprises and media companies. The messages are generally used for mobile marketing, enterprise and entertainment. However, banks often use them for fraud control. For example, if criminals are circulating a fake email that is asking people who have accounts at a certain bank to provide their social security numbers or other confidential information, these text messages can alert people to the scam so they do not fall victim to it. Bulk messaging is often utilized for reminders and alerts. However, it is more frequently used to send communications and information between customers and staff of various companies. Bulk messaging enables the delivery of SMS messages to large numbers of mobile phones that are located all around the world.

    Bulk messaging software

    In order to receive and send bulk messages, software is needed. There are many types of software packages specifically designed for this task that are available. These packages give their users the ability to send messages to as many phone numbers as they want. There are many different ways in which these phone numbers can be managed.

    The vast majority of software applications that are designed to be used with SMS enable the user to upload mobile phone number lists with the use of a CSV or TXT file. Systems that are more advanced are capable of automatically deleting any numbers that are repeated. There are also systems that can be programed to validate all of the mobile phone numbers before the messages are sent to them.

    Enhanced software features are also currently available that allow users to schedule messages to be delivered at certain days and/or times. Bulk messages are also able to be sent on mobile networks that are international or national, assuming that the provider of the bulk messaging software sends internationally.

    Bulk messaging portal

    Bulk messaging features can be added to websites through the use of this specific online script. Unlimited mobile phone numbers can be added to the list of numbers to send messages to. There are a wide variety of ways that can be used to manage these numbers.

    Bulk messaging API

    The majority of services that handle bulk messaging use the API's (Application Programming Interface) listed below. These enable the addition of functionality to programs by their programmers:

    • Email
    • HTTP
    • SMPP (Short Message Peer to Peer)
    • FTP (File Transfer Protocol)

    Immediate benefits of using bulk SMS messaging

    When a particular business is not doing well financially, they need to utilize various tools that can help them gain a competitive advantage in their specific industry. One of the main reasons that bulk SMS is so popular is its ability to lower operational costs while also generating revenue at the same time. Bulk SMS might be the only medium that is able to show a return on investment that is able to be measured. Wholesale SMS messaging is targeted, which makes it extremely effective at getting people to respond and generate revenue.

    Reduces operational costs

    Bulk SMS message transmission is more effective than email and less expensive than voice calls. There are thousands of businesses located all around the globe that utilize wholesale SMS as a way to communicate with their suppliers, employees and customers. There is a significant cost savings as a result of time being saved because actual voice calls to suppliers, employees and customers do not need to be made. A single message can instantly be sent to many people at the same time, as long as the person is located in an area with mobile coverage. The ability to disseminate information so quickly to a large target audience reduces communication costs while also generating revenue if used for marketing purposes.

    Allows customers to be accessed easily

    More people have mobile phones than have access to email or landline phones. Every mobile phone supports the use of text messaging. All mobile phone users are comfortable using this technology because it is simple and easy to understand. This makes wholesale SMS the perfect medium to use for communication with customers. There are also no demographical or geographical restrictions. ...

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  • 05/25/15--16:59: 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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  • 05/25/15--17:06: 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


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  • 05/25/15--17:08: New Software Releases
  • This page is to inform on various VoIP related software releases.

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


    May 2015


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  • 05/27/15--00:05: Bandwidth consumption
  • VOIP Bandwidth consumption naturally depends on the codec used.

    When calculating bandwidth, one can't assume that every channel is used all the time. Normal conversation includes a lot of silence, which often means no packets are sent at all. So even if one voice call sets up two 64 Kbit RTP streams over UDP over IP over Ethernet (which adds overhead), the full bandwidth is not used at all times.

    A codec that sends a 64kb stream results in a much larger IP network stream. The main cause of the extra bandwidth usage is IP and UDP headers. VoIP sends small packets and so, many times, the headers are actually much larger than the data part of the packet.

    IAX2 trunking helps with the IP overhead, but only when you are sending more than 2 or so calls between the same Asterisk servers. John Todd has done some useful practical testing, named IAX2 trunking: codec bandwidth comparison notes and results.

    The bandwidth used depends also on the datalink (layer2) protocols. Several things influence the bandwidth used, payload size, ATM cell headers, VPN headers, use of header compression and IAX2 Trunked. You can see the influence of some of this factors using the Asteriskguide bandwidth calculator.

    Teracall has the table which shows how the codec's theoretical bandwidth usage expands with UDP/IP headers:

    Codec BR NEB
    G.711 64 Kbps 87.2 Kbps
    G.729 8 Kbps 31.2 Kbps
    G.723.1 6.4 Kbps 21.9 Kbps
    G.723.1 5.3 Kbps 20.8 Kbps
    G.726 32 Kbps 55.2 Kbps
    G.726 24 Kbps 47.2 Kbps
    G.728 16 Kbps 31.5 Kbps
    iLBC 15 Kbps 27.7 Kbps

    BR = Bit rate
    NEB = Nominal Ethernet Bandwidth (one direction)


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  • 05/27/15--05:20: AX-E-1600P
  • Asterisk cards AX-E-1600P

    16 ports FXO/FXS card for Asterisk PBX

    AX-E-1600P Asterisk card is the telephony PCI card that support up to sixteen FXO and FXS ports
    Using AX-E-1600P analog card, open source Asterisk PBX and stand alone PC,
    users can create their SOHO telephony solution include all the sophisticated features of traditional PBX, and extend features such as voicemail in IP PBX.
    The FXO and FXS cards are interchangeable to suit various requirements.

    Features:

    • Analog card for Asterisk PBX
    • Support Asterisk PBX and zaptel/Dahdi driver
    • Support up to sixteen fxo/fxs analog port
    • Caller ID and Call waiting Caller ID
    • Conference

    Application:
    • IP PBX for Small and Medium Business
    • Call Center solution
    • PSTN trunking gateway

    Configuration:
    • Module:
    • Motherboard: AX-E-1600P
    • Dual channel FXS module: AX-210S
    • Dual channel FXO module: AX-210X
    • Hardware Requirement:
    • 500-Mhz Pentium III
    • 64MB RAM
    • PCI-Express interface
    • AX-E-1600P analog card
    • AX-210X dual channel FXO module
    • AX-210S dual channel FXS module

    Accessories

    • 1 RJ45 to 4 RJ11 port converter

    Contact ATCOM:

    ATCOM Technology co., LTD.
    Address: A2F , Block 3 ,Huangguan Technology Park , #21 Tairan 9th Rd, Chegongmiao , Futian District , Shenzhen China
    Tel: +(86)755-23487618
    Fax: +(86)755-23485319
    E-mail: sales@atcomemail.com
    Website: http://www.atcom.cn

    European Distribution:

    maple4VOIP | ATCOM Authorized Channel Partner
    Alsacecom | VoIP Product Reseller - France and EU Distribution.
    IPChitChat UK and European Distributor for Atcom cards and Atcom AXE1600P
    VoIPtel We ship worldwide


    support forum

    bbs.atcom.cn

    See Also





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  • 05/27/15--05:21: AX1600P
    • 16 FXO/FXS Analog Asterisk Telephony Card, PCI interface

    Image

    AX1600P is a telephony card working on open source Asterisk IP PBX system which supports 1 to 16 FXO or FXS ports, the ports can be configured by different combination of modes on the mainboard. Users can use AX1600P to install open source and powerful Asterisk IP PBX system.

    Features:

      • Support Asterisk ,Freeswitch , Dahdi , Zaptel
      • Support Trixbox , Elastix , Askozia
      • Validated by Elastix
      • PBX/Voicemail/IVR/Call Center/Call Park/Call Pickup/Call Transfer/Call Forward/Caller ID/Call Waiting/Call Conference

    • Module configuration
      • Slots for modules : 8
      • Dual port FXO: AX210X
      • Dual port FXS : AX210S
      • Dual port FXO/FXS: AX210XS





    Contact ATCOM:

    ATCOM Technology co., LTD.
    Address: A2F , Block 3 ,Huangguan Technology Park , #21 Tairan 9th Rd, Chegongmiao , Futian District , Shenzhen China
    Tel: +(86)755-23487618
    Fax: +(86)755-23485319
    E-mail: sales@atcomemail.com
    Website: http://www.atcom.cn

    European Distributors:

    maple4VOIP | ATCOM Authorized Channel PartnerATCOM Authorized Channel Partner
    Alsacecom | French VoIP Product and Service Reseller - France and EU Distribution.
    IPChitChat UK Distributor for Atcom asterisk appliances and new Atcom IP2G4A
    VoIPon - UK & Worldwide - Best Prices and Support on Atcom IP04 - Call for reseller pricing or International Shipping. VoIPon.
    uCpbx

    French Resellers:

    Alsacecom | French VoIP Product and Service Reseller - France and EU Distribution.
    Neotiq Support and expertise in French on the products.

    Australasian partners:


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  • 05/27/15--05:25: 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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  • 05/27/15--05:31: Sangoma
  • logo.gif


    Connect with Sangoma

    Sangoma is a leading provider of hardware and software components that enable or enhance IP Communications Systems for both telecom and datacom applications. The product line includes data and telecom boards for media and signal processing as well as gateway appliances and software.

    Compatible with Asterisk, Freeswitch, CallWeaver, Yate, 3CX, pbxnsip and others.

    VoIP Gateways

    Session Border Controllers

    Microsoft® Lync™ Solutions

    Multiplexers

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  • 05/27/15--05:35: beroNet GmbH
  • logo.JPG

    BeroNet is a German company founded in 2002 by 3 engineering entrepreneurs and is a recognized expert in the development of products and technologies for both the enterprise and service provider markets that enable reliable and efficient VoIP, ISDN & GSM connectivity technologies and through access devices engineered and manufactured in Germany.

    BeroNet is emerging as the leading designer and manufacturer for low density, high density & GSM VoIP access technologies through PCI Cards and Gateway products that are gaining both in dominance and preference by a growing number of PBX, OpenSource, Call Center and IVR integrators as well as PBX equipment manufacturers and operators.

    BeroNet is dedicated to bringing the world of advanced VoIP telephony solutions to allow the evolution and embrace of next generation of IP communications with technologies that combine the best of the Internet with the best of the existing fixed and wireless telephony infrastructure, breaking down the barriers to communications and making it easier for people and businesses everywhere and anywhere to communicate.

    Website: http://www.beronet.com
    Shop: http://shop.beronet.com
    Sales: sales@beronet.com
    Support: support@beronet.com
    Tel : +49 (0)30-259389-0
    Fax: +49 (0)30-259389-19


    Products:

    beroNet Telephony Appliance

    beroNet_Appliance_w300_schatten.jpg
    The beroNet Telephony Appliance is the ideal platform for customers and technology integrators, looking for a reliable hardware solution, with integrated ISDN, Analog and GSM connectivity.



    beroNet VoIP Gateways

    beroNet_Gateway_w300_schatten.jpg
    The berofix Gateways are modular where you can plug and mix different access modules in analog (FXS/FXO), digital (BRI/PRI ISDN) and GSM all on the same box.



    beroNet VoIP Cards

    beroNet_Card_w300_schatten.jpg
    the berofix Cards aren't the typical PCI/PCIe cards for which you would need proprietary drivers. Because of it's hardware design, the operating system detects the berofix PCI/PCIe Card as a standard network card. All necessary drives are automatically loaded by the onboard OS.



    Resellers and Distributors


    EUROPE

    • France

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  • 05/27/15--05:42: Openvox
  • Image

    http://www.openvox.cn

    Manufacturer of asterisk hardware products includes:


    Analog Cards:
    • A400P Series 4-port FXO/FXS PCI card (compatible to TDM400P)
    • A400E Series 4-port FXO/FXS PCI Express card
    • A400M Series 4-port FXO/FXS Mini-PCI card
    • A800P Series 8-port FXO/FXS PCI card
    • A800E Series 8-port FXO/FXS PCI Express card
    • A810P Series 8-port FXO/FXS PCI card with Lifetime Warranty
    • A810E Series 8-port FXO/FXS PCI Express card with Lifetime Warranty
    • A1200P Series 12-port FXO/FXS PCI card
    • A1610P Series 16-port FXO/FXS PCI card with Lifetime Warranty
    • A1610E Series 16-port FXO/FXS PCI Express card with Lifetime Warranty
    • A2410P Series 24 -port FXO/FXS PCI card with Lifetime Warranty
    • A2410E Series 24-port FXO/FXS PCI-E card with Lifetime Warranty

    ISDN BRI Cards (with Lifetime Warranty):
    • B100M 1-port BRI Mini PCI card
    • B100P 1-port BRI PCI 5V/3.3V Universal Card supports bri-stuff and mISDN
    • B100E 1-port BRI PCI Express card
    • B200P 2-port BRI PCI 5V/3.3V Universal Card supports bri-stuff and mISDN
    • B200E 2-port BRI PCI Express 1.0 card
    • B200M 2-port BRI Mini PCI card
    • B400P 4-port BRI PCI 5V/3.3V Universal Card supports bri-stuff and mISDN
    • B400E 4-port BRI PCI Express 1.0 card
    • B400M 4-port BRI Mini PCI card
    • B800P 8-port BRI PCI 5V/3.3V Universal Card supports bri-stuff and mISDN
    • PFM100 Power Feeding Converter for ISDN BRI Mini Card

    T1/E1 PRI Cards:
    • D110P 1-port T1/E1/J1 PCI Card(compatible to TE110P)
    • D110E 1-port T1/E1/J1 PCI Express 1.0 card
    • DE115P 1-port T1/E1/J1 PCI Card with hardware echo cancellation module EC100-32
    • DE115E 1-port T1/E1/J1 PCI Express Card with hardware echo cancellation module EC100-32
    • D210P 2-port T1/E1/J1 PCI Card(compatible to TE210P/TE205P), PCI 5V/3.3V Universal Bus Master
    • DE210P 2-port T1/E1/J1 PCI Card with hardware echo cancellation module EC100-64
    • D410P 4-port T1/E1/J1 PCI Card(compatible to TE410P/TE405P), PCI 5V/3.3V Universal Bus Master
    • DE410P 4-port T1/E1/J1 PCI Card with hardware echo cancellation module EC100-128
    • D210E 2-port T1/E1/J1 PCI Express 1.0 card
    • DE210E 2-port T1/E1/J1 PCI Express Card with hardware echo cancellation module EC100-64
    • D410E 4-port T1/E1/J1 PCI Express 1. ...

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  • 05/27/15--13:12: 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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  • 05/27/15--14:02: VoIP Wholesale
  • Wholesale VoIP Market:


    There is no doubt today that VoIP is taking over the telecom market, and every month increases penetration into services and industries. Competitive carriers are looking at the numerous ways to make money from this exploding technology, but there's a lingering question as to whether it is profitable to deliver VoIP in a wholesale model? Their customers, typically Service Providers, are looking for their ‘competitive advantage' into this ‘lowest price' race, leveraging within three key alternatives for packet telephony : “build” , “buy” or “rent”. Business aspect, there’s no need to invest tens of millions of dollars in wholesale VoIP to join in. Many Telecom Companies have done the work for you. They offer a complete, turnkey VoIP business service and equipment. Now you can start wholesale VoIP business with virtually no investment and yet reap great dividends.

    Wholesale VoIP Resellers:


    In today’s world, Service providers seeking to deliver VoIP to as wide a customer base as possible may find that becoming a wholesale VoIP reseller is the way to go. Wholesale VoIP may be sold to both other service providers and to enterprises or residential customers.

    Reselling IP telephony as a wholesale VoIP company is becoming an increasingly popular business model. For many companies, becoming a wholesale VoIP provider hits the sweet spot between profit and market control. Any firm with a well-established customer base is a good candidate for reselling wholesale VoIP.
    Becoming a wholesale VoIP reseller is not a decision that should be taken lightly. It does, however, offer the potential of being very lucrative if done right.

    Wholesale Consumer Demand:


    An important characteristic of the industry is the complex segmentation of consumer demand and rapid change in the characteristics that are being demanded, both at the end customer and in the intermediate ones (wholesale customers).
    Demand coming from ‘packed customers'? will be significantly different of the conventional telecommunications one, were telephony was the unique service to provide and differentiation was based on tariff-distance paradigm, being today's service offerings closer to data applications rather than telephony. Voice communication (and not old POT telephony) becomes the common feature into several communications applications and devices, but not the unique one.
    Messaging, conference, collaboration, web contact centres, etc … requires a common communication format between parties, which is voice, implemented through VoIP technologies. Heterogeneous and rapidly changing customer demands and products are important dynamic influences on the evolving structure of the telecom industry, resulting into a new value-chain.
    Telecommunication markets evolution will be driven by ‘packed customers' demand rather than networks, technology or finance, changing many decades rules into this industry.

    Finance in Telecommunication Industry:

    Finance institutions had been influencing Telecomm Industry since the beginning, due the business itself was characterized by huge investments, big market shares and bigger capitalization, influencing in many cases top management, who addressed their strategy towards ‘stock' opportunities rather long term and solid business models. WorldCom crash has been an example of this ‘financial market' pressure and wrong business management.
    Today, the networks has been deployed. New scenario in Telecoms enable new players to deploy services over broadband without proprietary network and this new generation business will not be anymore capital intensive, let's say these will be innovation intensive.

    U.S. VoIP Market:

    The US market for VoIP advanced dramatically in 2006-2007, adding 3.8 million VoIP households in 2006, reports In-Stat: As a result, wholesale VoIP revenues grows quickly, as MSOs, Skype, and a myriad of new entrants most lacking network facilities enter the market and drive demand for telephony features and applications, the high-tech market research firm says.
    As retail VoIP expands, wholesale VoIP will accelerate quickly, says Bryan Van Dussen, In-Stat analyst. The largest segment remains international VoIP, but we expect the market for local services to surge from 12% of all revenues to 27% by 2010.
    Recent research by In-Stat found the following:

    • Consumer VoIP adoption will drive wholesale VoIP revenues to $3.8 billion by 2010 from $1.1 billion in 2006.
    • In-Stat finds small businesses are driving the growth of hosted services in the U.S. Hosted VoIP seats in the U.S. ...

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  • 05/27/15--14:02: VOIP Resellers
  • VoIP Resellers


    This is a list of VoIP resellers in Alphabetical Order:

    1Pipe Telecom | OnePipe is a CLEC providing SIP Trunking, Hosted PBX, IP and other services, we provide services on our own network and gateways.
    • Custom portal for your customers
    • Actual US CLEC
    • Set your own pricing for your customers

    8774e4voip.com 8774e4voip.com - Contact Us Toll Free @ 877.434.8647

    Air21group.co.uk - UK BASED/ 247 SUPPORT - Call us on 0121 314 1114
    • Add to your telecoms business or start your own business reselling business telecoms today with the Air 21 Group
    • Whitelabel reseller program
    • High Commissions
    • Wholesale VoIP origination
    • Termination SIP Trunking
    Click to enquire now

    Alcazar Networks Inc. is offering Wholesale Origination / Termination / Free Toll Free Termination - Get paid for your toll free traffic! We provide high quality, dependable access to over 3100 rate centers and instant access to over 1,200,000 DIDs including T38 and local number portability. Wholesale SIP

    RingRoost - Enterprise level hosted PBX for companies and resellers world wide, featuring an easy to use PBX & call flow system builder.
    • Free SIP Inbound & Peer to Peer calling
    • Local Phone number 60¢/month
    • 2¢/min outbound calls and 0.7¢/minute inbound
    • Call Queues, Database Integration, IVR’s, PBX’s, CNAM Lookups
    • E911 Enabled
    • Toll Free Numbers
    • Free $10/credit trial version



    businessphonereseller.com Call us at 1-866-553-3833
    • Your branding
    • Business plans start at $10 per seat per month (wholesale)
    • You set end user prices
    • All features included
    Click to inquire


    • Smart Voice Network offering service to Carriers and Call Centers with unique ability to cap rates and manage sub accounts. VoIP Carrier with US & International Canada $.009 UK, France 0.0082,Mexico $0. ...

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  • 05/27/15--14:02: Sip Trunking Providers
  • This page is a list of SIP trunking providers around the world. Please keep this list in alphabetical order. SIP providers looking to add their services can do so in the list below.

    Country specific pages:


    What Is SIP Trunking?

    Traditional phone calls go over phone lines, but with the rise of the internet, Voice Over IP communication systems give you much more functionality to receive a high volume of calls. This is called SIP (session initiated protocol) trunking. These services are offered by the majority of VoIP providers and can be tailor made to suit the needs of a small business all the way up to a large corporation with a gigantic workforce and everything in between.

    One of the main advantages of going with SIP trunking is that it will integrate everything from data, the internet, your video, and your voice services all into a single line. The term "trunking" is a bit dated and goes back to a time when traditional telephone services were used and it described how a wide variety of telephone users could share a much smaller pool of communication paths, thus making it a much more efficient option for businesses than getting a separate phone line for each worker. That was in the past, though, as today's SIP trunking capabilities will even allow all calls to be considered local calls and won't even require the business to have a single phone line at all.

    The Benefits of Using SIP Trunking Services

    Choosing to use a good SIP trunking service will significantly level the playing field for smaller business to compete with large corporations and appear much more professional and bigger than they actually are. For larger business, one of the main advantages is being able to much more easily maintain that local presence. Regardless of the size of the business itself, SIP trunking's biggest benefit overall is simply the reduction in cost to the company's overall expenses. This is particularly true for SIP VoIP services that are specifically crafted for a business in particular.
    • It offers very low cost calling.
    • It's much easier to scale than other options, making it very future proof.
    • SIP trunking makes moving a breeze, as you can easily take your phone number with you while moving offices or even while traveling.
    • Network outages are much less impactful, as incoming calls can easily be routed to other locations.
    • It's ideal for any sized business with at least 25 physical phones.
    • It's a fantastic choice for any business that has an international location.
    • It offers the same sort of emergency usages and will even offer enhanced 911 dialing options.

    How SIP Trunking Can Take Your Business To The Next Level

    It used to be that you had to have a PRI (Primary Rate Interface) in order to have video, voice, and data all on a single line. This was a physical device that had to be purchased separately but along with your chosen phone provider's calling plans. This is no longer the case with SIP, as it allows your provider to set up a proxy server (commonly referred to as a SIP proxy), which will provide the outgoing, incoming, and PBX calls for you. Everything is located off site and handled by a professional IT team or contractor through your provider, meaning it saves you a ton of time and money on the technical support side of the equation.

    Being temporarily shut down means simply having the SIP reroute calls to a different office and also allows you to much more easily outsource to international locations, such as with customer service centers.

    All of these benefits are much better than what traditional PSTN (public switched telephone network) have been able to provide through their services and will give your business just what it needs to take it to the next level.

    Zoom Soft is worldwide

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  • 05/28/15--07:10: New Software Releases
  • This page is to inform on various VoIP related software releases.

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


    May 2015


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    Business VoIP Providers - Compare and Choose a Business VoIP Provider


    Quality business VoIP providers today offer a wide variety of feature packages, services and prices. Selecting the ideal provider and service options will depend on your type and size of business, features needed and projected volume of usage. Even when working with top-tier providers, your basic monthly service charges per line may begin at rates as low as $20. Before choosing your VoIP provider, it is essential to first determine your company's precise telecommunications needs to enable timely and cost-efficient initiation of your service. By consulting your chosen Voice over IP service team and seeking their expert advice in advance, you can be prepared to take the following steps to facilitate the smooth, productive startup of your services:

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

    Important Information to Request from Any Potential VoIP Provider

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

    Take Full Control and Advantage of Your VoIP System

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

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

    Major Business Benefits and Advantages of Installing VoIP

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

    • Schedule Your Own Business Hours. ...

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  • 05/28/15--16:39: 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


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