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VOIP Billing

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Hosted Billing Services (in Alphabetical Order).

  • 4PSA VoipNow fully featured, carrier-grade, multi-tenant edition for service providers and businesses, that can be installed on their chosen infrastructure or delivered as a UCaaS. VoipNow provides a fast, competitively priced go-to-market solution, from deployment and provisioning all the way to selling and billing.
  • Avatar Dialler Make your billing procedure easy with easy voip billing! avatar is offering you free voip billing software where you can also check your balance at any time.
  • Adore VoIP Billing Adore VoIP Billing Software comes with the enhanced functionality along with the architecture with class. It is fully compatible and gets integrated with all other VoIP related products. It is designed with all the present and future demands of booming telecom industry kept in the mind. The telecom industry is changing and developing with rapid speed and the VoIP products such as VoIP Billing comes as an excellent product in this time.
  • A2BILLING - VoIP Billing Solution / AAA / Class 5 Softswitch.
  • Aradial AAA for Billing Solutions
  • Billingbooth - Telecoms Billing Fully featured Telecoms and CDR billing service, including support for multiple Telco Providers, Minute Bundles, Recurring line charges, rate sheets, analytics, customer area and direct debit integration.
  • Billrun - Open Source Billing, Designed for Big Data BillRun is an Carrier Grade, X-Play Billing system (X = fixed line, Mobile, Broadband, VoD, IoT, etc.) implemented as a Service (SaaS) or on-premise. It was designed and developed with the latest technologies to allow you, the customer with total peace of mind. Boost your business utilizing BillRun’s unique cloud-based open-source affordable billing solution, providing you with the tools you need for making real-time and future decisions. PAY AS YOU GROW - Secure your business in the long term with an affordable billing solution, and avoid the risk of over-investing during marketplace uncertainty. RESPOND INSTANTLY - Gain the agility to react immediately to market demand, by creating a campaign, automatically calculating its costs, and launching it within minutes. INCREASE REVENUES - Create personalized promotions that improve loyalty and stimulate your customers to purchase more services and products. SCALE OUT - Enjoy peace of mind knowing that the same high-speed performance and flexibility are guaranteed even when your customer base increases significantly.
  • benotos offers free callshop billing system 4-level billing system: reseller-subreseller-callshop-customer, 2 different routes, nice easy to use interface, intelligent ratemanager, online payment, detailled reports, receipt printing with own logo, white labelled, use your own brand and domain name and much more features. About 9000 callshops around the world are using our excellent callshop billing solution already. Free signup - best rates on market - low payment amounts
  • BillCall - Telecom Resource Management for wholesale Voip Carriers Panamax’s Telecom Billing Solution BillCall provides solutions for End-User billing, Carrier Access Billing (CABS), CDR Mediation, Rating & Routing.
  • CloudAstrix SPE CloudAstrix SPE is such a VoIP Switch. Build on the world renowned WHMCS Billing Suite, the Soft-switch module brings all necessary functions to perform and provide a top class VoIP service. ...

Asterisk Realtime ODBC

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asterisk version OS/Distro unixODBC threading pooling odbc-connector SQL server status
13.9.1 Linux/Centos 6 2.3.2 (backport fc20) 0 yes 5.1.5 distro MySQL 5.1.73 distro random segfaults
13.9.1 Linux/Centos 6 2.3.4 (backport fc23)0 yes 5.3.6 oracle rpmMySQL 5.1.73 distro OK
13.9.1 Linux/Fedora 23 2.3.4 distro0 yes 5.3.6 distro MariaDB 10.0 distro No Crash, but unixODBC locks



VOIP Service Providers Business Europe

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This is a list of VOIP Service Providers who offer full service products primarily aimed at the small to medium sized business telephone market. Such companies typically support multi-line telephone systems, small PBX gateways and hosted VoIP (as an alternative to Centrex service). See also:


Service providers operating in more then one country are listed under each country. "Operating in a country" means a provider that has billing and support staff located in the country, and offers service subject to the regulations and consumer protection laws of that country.

Please describe services in neutral language and normal fonts. Don't bother listing prices--unless you really plan to return and edit them as things change. If you want to add your company, please read the Posting Guidelines for Promoting Products and Services. When you add your entry to this page, please make sure your entry is in alphabetical order in relationship to other vendors listed in the same section. Failure to follow these guidelines will result in deletion!


Please include relevant information like SIP or IAX handoff, how outgoing CLID is set, whether a given account may originate multiple calls at once, etc.

Users: Please feel free to REMOVE any listing that does not meet the stated goals of this page. ...

VOIP Service Providers

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For a list of VOIP to PSTN service providers, indexed by country, please see:


VoIP and VoIP Service Providers

What is VoIP?

VoIP (which stands for "voice over internet protocol" and is commonly referred to simply as an internet phone) is a highly cost effective and reliable way for businesses or even homeowners to make calls across the world or even just across town. The majority of major cable companies that offer bundled internet, television, and phone services already utilize this newer technology, but there are tons of other independent companies that specialize in providing this service to their customers at reasonable rates and with tons of extra features.

More Than Just Computers

When people think of VoIP, they generally think of computers due to the popularity of the numerous free communication services like FaceTime and Skype, but this is truly just one aspect of what VoIP can truly offer. It is true that VoIP technology transmits voice communication that's been converted into digital data across a packet-switched network or the internet (what this means, in essence, is that a user making phone calls over high speed internet lines rather than phone lines). With that in mind, users are not confined to only using it on a computer. VoIP technology can connect through the internet using traditional telephone equipment just like a regular line. The phone itself is connected to the internet using an adaptor that's plugged straight into a home or business's internet network. Most major services offer a softphone option as well, which allows the user to use their computer directly as a telephone service. ...

Android VOIP

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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

Apple iPhone VOIP

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IVR

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What Is IVR?


IVR, or interactive voice response, is a what allows phone systems to process touch tones or voice waves during a telephone call. IVR technology is responsible for the menus people hear and respond to when they call up a company or business and hear the words: "press 1 for sales, press 2 for marketing, press 0 to speak to the operator," for example. IVR systems can be fully customized to play back dynamic audio, or pre-recorded menu options.

IVR is not necessarily related to VOIP, however, a VOIP IVR is. Most VOIP IVR systems or software support SIP based VOIP, but Skype IVR also support non-standard based Skype service.

Computer Telephony Component

IVR is an automated computer telephony integration CTI system which allows providers to create complex menus which the caller can navigate by using touch-tone key-presses or via spoken commands. IVR systems can be used as a Voice portal to access remote information such as bus scheduling where the caller can select the route for which they require information, or for billing or customer service systems which allow the caller to enter information such as their account number or credit card details without the need for operator assistance.

IVR and ACD Integration

IVR solutions are often integrated with an ACD, which routes incoming phone calls to agent work groups. This integration can be both a front end and back operation.

  • Most typically, an ACD system can route callers to an IVR program based upon DNIS or other parameters such as time of day or day of the week.
  • A smart IVR can transfer callers back to an ACD system to route the call to the next available agent within an agent hunt group.

One important task of an integrated IVR and ACD is to display Screen Pop information from the caller on the agent's workstation so that the agent has caller information readily available without the need to prompt the caller again.

IVR and Voice Broadcasting

IVR applications are typically associated with inbound calling programs. However, IVR technology can be applied to outbound calling campaigns and are most commonly used with Voice Broadcasting and touchtone responses. Examples of the application of this technology include the option to speak with an operator, opt out of a calling campaign, or taking an outbound survey.

Graphical Design Tool for IVR Applications

Recent IVR systems usually use high level scripting languages such as VoiceXML, an open standard for interactive voice response systems. For most users who lack technical training, developing an IVR system using scripting language, even high level language, are not feasible. The good news is there are design tools that are based on graphical user interface for the techies and none-techies alike. By using a GUI tool, a user can simply drag-and-drop components and create and deploy an IVR system in minutes. The whole design is a call flow diagram, much like a voicemail system user manual.

See Also (Vendor Information)

IVR Information

  • http://www.evoxtel.comEvoxtel Intelligent IVR ,Automatic call distributor ACD,Call Forwarding,call SMS back
  • CCXML standard markup language for IVR / call control applications
  • IVR System Simulation Model - estimates resources required for an inbound calling campaign.
  • IVRS World - Blog about IVR

VoIP IVR Systems

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VOIP IVR Systems combine VoIP technology with IVR (Interactive Voice Response) automatic call answering software.

See Also (Vendor Information)


The following is a list of vendors that provide interactive voice respone (IVR) phone systems and software solutions:


Hosted IVR Solutions

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This page lists providers of Hosted IVR solutions.

Hosted IVR Solutions Providers


Acclaim Telecom Services, Inc.

1300 E. Arapaho Richardson, Texas 75081
Suite 206
Richardson, Texas 75081
866.324.6416

Acclaim Telecom delivers the following solutions for your IVR needs:

  • IVR Hosted Solutions / Software as a Service (SaaS) / Cloud Computing Solutions
  • Custom IVR Application Development and Support
  • IVR Managed Services / Managed Premise IVR Solutions
  • Acclaim Telecom Mobile Solutions
  • IVR Extended Life Maintenance Programs

We Extend the Value of your IVR Systems

It’s not just about technology; it’s about solving your business needs. Acclaim Telecom Services helps our clients achieve greater top line revenue, improved customer satisfaction, increased company productivity, and reduced operating costs.

For more info: http://www.acclaimtelecom.com
For more info details on IVR Solutions:
IVR Hosting - http://www.acclaimtelecom.com
Hosted IVR Solutions - http://www.acclaimtelecom.com
IVR solutions - http://www.acclaimtelecom.com/solutions/



Arcutel - Enterprise Hosted IVR

https://www.arcutel.com

We provide an automatic telephony service to businesses of all sizes. You setup and have full control over your own IVR via our web based application. There is no investment required, we don't charge extra for premium features, and the first month is free.



AVOXI from AVOXI


AVOXI virtual call center solutions provide virtual call center products like SIP trunking and VoIP gateway solutions, with international toll-free numbers. Contact AVOXI at +1-770-937-9735 to learn more about these products and services.


Other Services:
Cloud Based Contact Center
Freephone Numbers
Hosted PBX Providers
International Toll Free Number
VoIP Call Center

Bright Pattern Interactive Voice Response


Bright Pattern delivers next generation cloud contact center solutions, including Interactive Voice Response. Gracefully build your IVR logic flow using our drag-and-drop interface, Scenario Builder, part of ServicePattern, our platform. Quickly build sophisticated interactive voice response applications that follow sequential or branching flows. ...

Bandwidth consumption

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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)

Hosted PBX Vs On Premise PBX

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There are pros and cons of both hosted PBX as well as on-premise PBX. There are some fundamental differences to each of the systems and they feature advantages that should be known prior to making a decision on one or the other. The move to an IP-PBX business phone system is beneficial regardless of which system is ultimately chosen. However, there are differences and knowing those leads to a better VoIP phone system and a higher level of satisfaction for the company, employees and even callers.

Open source on-premise systems, such as Asterisk, are responsible for driving down costs for VoIP providers and manufacturers. This provides users with the ability to get the latest technology with many more features at a lower cost than what was traditionally available.

What is Hosted PBX?


Hosted PBX or hosted VoIP, otherwise known as an Internet phone system is one where the provider is responsible for housing the IP-PBX as well as handling the technology required to provide the services to the phone system. The desk sets will plug into a router and the calls, signaling, and features are handled through an IP-PBX server at the provider’s location. The provider of the hosted PBX charges a monthly fee that is inclusive of a minutes package and potentially certain features. Charges can also be at a per minute calling cost. Either one can be affordable depending on the rates. A company that knows the amount of minutes spent on the phone in a given month can make effective cost comparisons. Extended features may come with additional cost.

What is On Premise PBX?


On-premise PBX is also known as an IP-PBX phone system. It is similar to a traditional PBX system that resides at a location, such as a computer equipment room or phone closet. The main difference is that IP routing is done with more current technology. The signaling is done with an IP phone to the IP-PBX server using a LAN. Calls can go through a traditional phone company as well as voice over Internet (VoIP) using SIP trunking. Gateway cards are used to connect the system to the traditional phone company provider. The provider can be the one that already provides service, though a SIP trunk can be configured for use with an Internet service telephone provider (ISTP). An Asterisk based system is the most affordable option for on-premise PBX due to the flexibility that is offered with open source software.

Hosted IP-PBX versus On-premise IP-PBX


There are some differences between the two options. Understanding benefits and limitations makes it easier to determine the best option for any particular organization. Cost, expansion, and other considerations are laid out to make it possible to compare the hosted IP-PBX and the on-premise IP-PBX within the same categories to learn of the greatest differences.

Costs


Purchasing an on premise IP-PBX phone system involves buying hardware, which includes a server with the proper number of interface cards (if needed) to be able to connect the telephone company with the IP phones. Hosted IP-PBX only involves purchasing IP phones, though a router and network switch may be needed to ensure there is one specifically dedicated to VoIP.

Hosted PBX:

  • Lower initial equipment cost and set-up cost
  • Network qualification is performed by the customer. ...

SIP

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SIP, the session initiation protocol, is the IETF protocol for VOIP and other text and multimedia sessions, like instant messaging, video, online games and other services.

Abstract from the RFC 3261 (formatted_and_explained version) - SIP: Session Initiation Protocol

This document describes Session Initiation Protocol (SIP), an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences.

SIP invitations used to create sessions carry session descriptions that allow participants to agree on a set of compatible media types. SIP makes use of elements called proxy servers to help route requests to the user's current location, authenticate and authorize users for services, implement provider call-routing policies, and provide features to users. SIP also provides a registration function that allows users to upload their current locations for use by proxy servers. SIP runs on top of several different transport protocols.


SIP is very much like HTTP, the Web protocol, or SMTP. Messages consist of headers and a message body. SIP message bodies for phone calls are defined in SDP -the session description protocol.

  • SIP is a text-based protocol that uses UTF-8 encoding
  • SIP uses port 5060 both for UDP and TCP. SIP may use other transports

SIP offers all potentialities of the common Internet Telephony features like:
  • call or media transfer
  • call conference
  • call hold

Since SIP is a flexible protocol, it is possible to add more features and keep downward interoperability.

SIP also does suffer from NAT or firewall restrictions. (Refer to NAT and VOIP)

SIP can be regarded as the enabler protocol for telephony and voice over IP (VoIP) services. The following features of SIP play a major role in the enablement of IP telephony and VoIP:

  • Name Translation and User Location: Ensuring that the call reaches the called party wherever they are located. Carrying out any mapping of descriptive information to location information. Ensuring that details of the nature of the call (Session) are supported.
  • Feature Negotiation: This allows the group involved in a call (this may be a multi-party call) to agree on the features supported recognizing that not all the parties can support the same level of features. For example video may or may not be supported; as any form of MIME type is supported by SIP, there is plenty of scope for negotiation.
  • Call Participant Management: During a call a participant can bring other users onto the call or cancel connections to other users. In addition, users could be transferred or placed on hold.
  • Call feature changes: A user should be able to change the call characteristics during the course of the call. For example, a call may have been set up as 'voice-only', but in the course of the call, the users may need to enable a video function. A third party joining a call may require different features to be enabled in order to participate in the call
  • Media negotiation: The inherent SIP mechanisms that enable negotiation of the media used in a call, enable selection of the appropriate codec for establishing a call between the various devices. This way, less advanced devices can participate in the call, provided the appropriate codec is selected.

The SIP protocol

The SIP protocol defines several methods. ...

New Software Releases

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Asterisk cmd MusicOnHold

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Synopsis

Play Music On Hold indefinitely.

Description

MusicOnHold([class])

Plays hold music specified by class. If omitted, the default music source for the channel will be used. If you have configured MusicOnHold in musiconhold.conf it will get played automatically if the extension is put on hold. This command FORCES musiconhold music.

The default MusicOnHold class is set with the SetMusicOnHold command (Deprecated in 1.6).

Example

Extension defined in extensions.conf with "forced" MusicOnHold. Remember to Answer before letting the music pour down the line. Otherwise music on hold will not work correctly.
; Answer required as Music On Hold does not answer the call
exten => 6000,1,Answer
exten => 6000,2,MusicOnHold()

It is often useful to turn off music on hold in several situations:
  1. when a particular extension calls (originates)
  2. when connectiong to a particular extension
  3. when traversing a particularly expensive network
  4. when connecting to a conference


(So, how do we do handle each situation?)

You can turn off MOH on a per call by using the SetMusicOnHold command (Deprecated in 1.6).

Add a new class to musiconhold.conf
[none]
mode=files
directory=/dev/null

Create a macro in extensions.conf to turn off MOH
[macro-nomusic]
exten => s,1,NoOp(Turn off MOH for this channel)
exten => s,2,SetMusicOnHold(none)

Now call this macro when you dial an extension
exten => 7020,1,NoOp(Dial -> IAX2/outbound/${EXTEN})
exten => 7020,n,Dial(IAX2/outbound/${EXTEN},,M(nomusic))
exten => 7020,n,Hangup

Asterisk 1.6

MusicOnHold(class[,duration])
Plays hold music specified by class. If class is omitted, the default music source for the channel will be used. Change the default class with Set(CHANNEL(musicclass)=...). If duration is given, hold music will be played specified number of seconds. If duration is ommited, music plays indefinitely.

Returns 0 when done, -1 on hangup.



play-fifo (3rd party addition)

This small C program will create if necessary, open and listen on a fifo for slinear audio and delivers it to STDOUT. If STDOUT is blocking, it discards the data. The idea is that you would use it in a custom class in res_musiconhold. Now you can use whatever means you choose in a seperate process to deliver raw 8khz mono slin to the fifo which will be heard as the music class fifo. An example would be to play your line-in into the fifo and the buffer will not overflow because this program does a poll on the STDOUT and discards STDIN when STDOUT is busy.
You can find it here, it is not part of the Asterisk distribution.

See also

Asterisk Appliances

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An Asterisk appliance is a computer with Asterisk installed or “embedded” along with an Asterisk GUI, and FXO and FXS ports. Generally, Asterisk appliances have smaller form factors than desktops or some server models, and are highly efficient at running an Asterisk PBX.

Asterisk appliances are useful for small businesses wanting a complete PBX that is efficient in size, performance speed, and energy usage. Asterisk appliances are less expensive than typical PBXs, ranging from $200-$1000.

Asterisk appliances can be custom-made or pre-built from proprietary sources, or a user can build their own. Because Asterisk can run on nearly any computer, there is a wide variety of choice when it comes to the individual components of an Asterisk appliance. For example, Asterisk can be configured, depending on the modules, for 32 or 64-bit architecture. Some Asterisk appliances can use netbook chipsets such as the Intel Atom for a smaller form factor.

When building an Asterisk appliance, compiling modules can restrict chipset and operating system. For example, the Astlinux project must use the uClibc compiler rather than the more common GCC compiler, which makes Astlinux incompatible with Digium’sG.729a codec.

Proprietary Asterisk appliances come in many forms and do not require the user to compile code. Rhino’s Ceros uses Intel Core i3, 160GB hard drive and 1GB RAM with whatever Asterisk distribution the user chooses. The Phonebochs Telephony Appliance from Rockbochs, on the other hand, uses Intel Core Duo Mobile and Trixbox.

Asterisk Appliance from Digium

Digium, Inc has its own product called Asterisk Appliance 50. The server is designed for small businesses (up to 20 users) and can function as a hybrid phone solution or complete VoIP PBX.

Other Proprietary Asterisk Appliances


Asterisk hardware

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GoForOpen

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High quality Application Appliance servers using Open Source software!



Our Products:

GoForOpen PBX Series - PBX50 / PBX125 / PBX500 IP PBX for Small to Medium Business


Ultra compact, feature rich PBX for every-day use. It's completely fanless and silent.


The GoForOpen PBX Series products are high quality IP Telephony appliances with all the basic features and some advanced features only found in large scale phone/ communication systems, but without the additional cost. The GoForOpen PBX Series products were designed to be highly reliable, flexible and easy to use for cost conscious small to medium sized businesses. Take advantage of all the big business features to impress your customers. Reduce your communications costs with VoIP.

Key Features and Benefits:

1. Models support concurrent calls and many extensions:
PBX50 - 50 concurrent calls and at least 200 extensions.
PBX125 - 125 concurrent calls and at least 500 extensions.
PBX500 - 500 concurrent calls and at least 2000 extensions.
2. Interactive Voice Response (IVR) and auto-attendant ensure all calls are answered 24/7 with the highly flexible IVR and auto-attendant even if no one is free to answer the call.
3. Voicemail - Configure a voicemail box for every extension without the additional fees.
4. Link extensions to any phone anywhere - Route calls to an extension to a cell phone, another office extension, a home phone or anywhere else to never miss a call when away from the desk.
5. Support multiple locations with trunks - Connect all your office locations together with a VoIP trunk to save on long distance charges.
6. Web-based configuration - Leverage all the features with the easy to use web-interface.
7. Conference bridge - More than a 3-way conference call, connect multiple participants to a virtual conference room with a professional IVR to handle in-coming calls from participants. Save on the costs of a conference call service with this free built-in feature.
8. Ring groups and hunt groups - Set up ring and hunt groups so that customers can reach your staff in the shortest amount of time.
9. Totally silent fanless chassis.
10. Power consumption < 10W.

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Call Center Solutions

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Call center solutions include any call center provider or kind of software that supports call centers. Call center solutions are products and services that solve the specific need for customer relationship management (CRM).

A call center solution is often a total package like unified communications or turnkey product, such as an outsourced call center. Software that runs a complete call center system would also be considered a call center solution.

Types of Call Center Solutions


Inbound call center solutions help companies with incoming calls, such as with customer service and tech support. Outbound call center solutions can help businesses with market research, lead generation and customer outreach. Telemarketers use outbound call centers.

Blended call center solutions can provide both inbound and outbound services.

A virtual call center solution can work for businesses with no central location or several locations. Call center agents can work from home.

Offshore call centers are outsourced companies outside of the business’ home country. Offshore call center solutions can save businesses on overhead expenses.

Web-enabled call center solutions allow customers to call in from their computers. A web-enabled call center is usually a contact center.

In many cases, call center solutions are actually contact centers with the ability to communicate in a wider variety than voice. See call center vs contact center.

Call Center Software

Software is always a part of modern call center solutions. A call center needs a server to run its phone system (such as a PBX) and software to integrate telephony and computer systems.
Not all software is all-in-one. For example, a company wishing to integrate CRM with its call center technology requires additional software.

Call center software can have useful features such as phone controls from the computer and screen popping a client’s profile/information.

Many call center services come with their own, proprietary software. When a company runs its own call center or contact center there are software solutions, both proprietary and open source, for the below functions:


Call Center Solutions

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

Android VOIP

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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

Apple iPhone VOIP

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