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Call Center Software

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Call center software is the software system that allows a company or organization to run a call center. This page lets you compare call center software providers.

There are hundreds of different providers of call center software across the globe, and every call center software system has its pros and cons. When selecting the right call center software for your business, contact center, or call center, it's important to decide which features you want your phone system to have.

Types of Call Center Software


ACD helps productivity by assigning inbound agents to incoming calls. The automatic call distributor uses a set of instructions to determine who gets the call in the system. The algorithm can route calls based on agent skill or whoever has an idle phone. ACD can use caller ID or automatic number identification, but usually interactive voice response is enough to help the system determine the reason for the call.

An automatic call distributor can also take advantage of computer telephony integration. Agents can receive relevant data on their computers along with the incoming call.

Computer telephony integration is a broad category of software that connects telephone and computer systems. Computer telephony integration software can have both desktop and server functions. Various applications make up a system that can control phones, display call information, and route and report calls.

Interactive voice response allows callers to route themselves to the appropriate department or use the company’s database for assistance. More sophisticated interactive voice response systems can access accounts and perform certain tasks, such as activating a credit card through a bank’s phone system. IVR involves using dial tone multi-frequency or voice commands. In the VoIP industry, a PBXauto attendant is near interchangeable with IVR. However, auto attendants are not capable of speech recognition.

A predictive dialer calls a list of phone numbers at once. Outbound agents are then connected to the numbers that answer. A predictive dialer uses calculations to minimize the idle time of agents and the potential of losing answered calls when no agents are available.

Contact Center Software

For contact centers, software includes applications for chat, email, and web interaction in addition to telephony functions.

Call Center Software Providers

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

  • 3CLogic Cloud-Based Contact Center Software
  • asterCC hosted Call Center Solutions
  • AVOXI Virtual Call Center Solutions

Advanced Network Devices

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Website: www.anetd.com
Support: www.anetdsupport.com
Forum: www.anetdsupport/forum.com

Products:

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Advanced Network Devices IPClock


Advanced Network Devices Power over Ethernet (PoE) Products That Easily Connect to Existing Data Networks:


IP Clock and IP Speaker are Power over Ethernet (PoE) synchronized clocks and intercoms that requires only an RJ-45 connector to connect to existing data networks. Simultaneously, broadcast to both phones and speakers. The clock auto synchronizes and can be used as a scrolling text display and a standard built in microphone allows two way voice communication. Works on existing data networks and inter-operates with Cisco and other VoIP phone systems. Full IP product line can be used together to meet specific needs. Send broadcasts to specific rooms, departments, or groups as needed. Bi-directional audio / intercom supported on most models.


For more information contact;
sales@anetd.com
847-463-2236

Technical support contact;
tech@anetd.com

Supported Software

IP Clockwise
Singlewire- Informacast
Syn-Apps- SA-Announce
IPcelerate- IPsession
Bell Commander
MessageNet

Where to Purchase

Predictive dialer

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What Is A Predictive Dialer?


  • "A predictive dialer is a computerized system that automatically dials batches of telephone numbers for connection to agents assigned to sales or other campaigns. Predictive dialers are widely used in call centers." - Wikipedia

"Definitions of Predictive dialer on the Web:

  • A predictive dialer is an outbound call processing system designed to maintain a high level of utilization and cost efficiency in the contact center. The dialer automatically calls a list of telephone numbers, screens the unnecessary calls such as answering machines and busy signals, and then connects a waiting representative with the customer.

note: above text may be copyrighted by it's respective owners)

A VOIP Predictive Dialer, a.k.a. soft predictive dialer, is a software product capable of predictive dialing using VOIP service directly. Besides computer and internet connection, there is no equipment needed in order to use VOIP predictive dialer.


Software Only Predictive Dialer

New predictive dialing technology, together with faster computers and bigger broadband bandwidth, enables software only predictive dialers to work as good as or even better than hardware based dialers. Software based solution avoids expensive telephony board and associated hardware maintenance cost. It is easy to install and configure. For example, it is very easy to setup remote agent (at home agent).

SOFTWARE ONLY PREDICTIVE DIALER
  • Altitude predictive dialer - from 10 seats upwards. http://www.altitude.com/en/products/automation/dialer.html Provides powerful predictive dialing solutions The solution intelligently automates the call to be done, the best time, number and agent according to priorities, skills, availability and contact strategy.
All pacing modes are available including preview, power and an accurate predictive dialer. Outbound contacts can be blended with inbound activities.
Contact us today and schedule a demo! info@altitude.com - 905-479-2655
  • 3CLogic Software Predictive Dialer - No hardware required. Unique software AMD algorithm eliminates dead air. Highly accurate with minimum wait time for agents. Agents can be working from home or multiple locations. Complete visibility into call center operations.
  • Avyukta Intellicall - Predictive Dialer + VoIP Minutes - 99 Live reference centers across the globe + Best Quality VoIP Minutes with redundancy so as to minimize downtime. Predictive Dialer + VoIP Minutes Free demo. IVRS, Domestic PRI Based Center, Bulk VoIP Termination. Various models for Dialer and VoIP as per your requirement. Contact: - Sukumar Singh, sukumar@dialerindia.com Message Me: - avyukta.sukumar @ Gtalk, Skype. +91-8560000080.
  • Genesis Communications predictive dialer - from 1 seats upwards. http://www.genesiscommunications.biz Provides Predictive dialing for at home agents as well as agents at single or multiple locations. Outbound, Inbound and Blended modes. Call Today for a demo or trial account! (949)276-4184 Also see our page on Voip-Info.org for more details: http://www.voip-info.org/wiki/view/Genesis+Communications,+Inc.
  • MDLsolutions,Inc. - VoIP Dialer and Predictive Dialer. ...

Asterisk Paging and Intercom

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Paging and Intercom

On legacy phone systems you can find the following kinds of paging:

  • Dial a code to connect to a separate overhead paging and announcement system (like in an airport)
  • Dial a code and connect directly to a built-in one-way announcement speaker on one or more phones
  • Dial a code and connect directly to a built-in two-way announcement and talkback function on one or more phones

Some overhead paging systems also provide a talkback system so that the person being paged can just speak to respond. Background noise issues limit where this feature can be used. The talkback function is usually setup to be hands free. That means that the person responding to the page does not need to take any action other then speaking.

New in Asterisk 1.2: The new dialplan command Page utilizes MeetMe to page one or more phones.

New in Asterisk 1.8: A new RTP engine and channel driver have been added which supports Multicast RTP.
The channel driver can be used with the Page application to perform multicast RTP paging. The dial string format is:

MulticastRTP/<type>/<destination>/<control address>

Type can be either basic or linksys. Destination is the IP address and port for the RTP packets. Control address is specific to the linksys type and is used for sending the control packets unique to them.

Advanced Paging / Intercom

There is also another system available since many years, the best one, combining paging and intercom. Here the talback system is limited to only one phone. The paging is done in one way mode through a group of phones, and the person being paged can respond pressing a digit to switch the nearer phone to two-way mode, simultaneously hanging-up all other phones speakers.
This mode combine the best of the two world, eliminate the noise problems, and keep the communication private as soon as the paged person pressed the right digit on a phone.
It should be possible to implement this mode on Asterisk with a managed conference and a feature map application.

Multicasting begin to be supported at all major phones manufacturers, Aastra firmware v2.2, Snom v7, Linksys,... allow the setting of a multicast listening address. This will permit to reduce the generated trafic for an extensive paging.

If a phone is in use when a page arrives, some systems can do a "whisper page" so that only the person being paged can hear the page.

SIP phones for the most part don't support any of these phone based paging functions. If a SIP phone offers an Auto Answer function, you can approximate limited paging intercom functionality. The phones most often mentioned supporting this are:

VOIP Phones

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This page is for listing brief details of VoIP Phones including details, where to buy, specifications, and any other relevant VoIP Phone information. Please read the Posting Guidelines for Promoting Products and Services before adding to it.

Hard Phones

Standalone Ethernet Hard Phones (voice only)
An Ethernet hard phone is a self contained IP telephone that looks just like a conventional phone but instead of a conventional phone jack, it has an Ethernet port through which it communicates directly with a VoIP server, VoIP gateway or another VoIP phone. Since a broadband hard phone communicates directly with a VoIP server, VoIP gateway or another VoIP phone it does not require any personal computer nor any software running on a personal computer to make or receive VoIP phone calls. It can be used independently, all that is required is an internet connection. While PC based software solutions are cheaper, a hard phone is the best solution for IP telephony.

General


VoIP Calling Card Platform Providers

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This page is for all of the different Calling Card Solution Providers out there that have Turn-Key Solutions to offer. Please post info about your solution with links to the Data Sheets or White Papers as well as a way for companies to contact you if they are interested.



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Aloha Wholesale


Free UK DIDs for calling cards. For customers who recieve more than 2,500/minutes a day we also pay out £0.001/min on calls received. Capacity is unlimited and welcome small and large calling card operators.

Contact us today.

Aloha Wholesale | Part of the Aloha Telecommunications Group, a UK National operator. Aloha Wholesale provide inbound (DID/DDIs) and outbound (UK National and A-Z Termination) services. As a UK national operator with our own independent network. Unlike other operators we own the resources we sell in regards to inbound (i.e all numbers are allocated to us via Ofcom the regulator). Aloha focuses on the quality end of the market rather than the LCR or grey route market. Services for Telecom Operators, Call Centres (inbound/outbound), ISPs, Mobile Operators and M2M. All inbound UK numbers (DIDs) contain unlimited channels as standard.


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Voice for Life with Caxtel Telecommunication Services


Caxtel Direct Call

From any phone set, you simply dial a local Caxtel direct number.
You can use Pin Number or save your Caller id in your account for authentication and then dial the international number you want to call.

Caxtel Web Call

From the web, you can initiate a call without headset, No download, No software installation required. Use any phone, start saving now!

  • Real-time rating, billing and routing.
  • Account balance monitoring.

Caxtel Telecom Limited.


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DIDForSale Provides the Complete CallingCard Platform

Front End Website Featuresl

  • Web Site Template
  • Online Signup
  • Display Different Rate plans
  • Access Numbers
  • Toll Free Numbers
  • Password Retrieval
  • Check Rates
  • Fraud Protection
  • Address Standardization & Verification
  • Referral Credit

Customer Portal

  • Payment History
  • Call History
  • Vouchers
  • Payment gateway Integration
  • Invoices
  • Pinless Dialing
  • Password Management
  • AutoRefill
  • Real-time rating, billing and routing.
  • Account balance monitoring. ...

DID Service Providers

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A Direct Inward Dialing service provider delivers a telephone number over VoIP protocol including SIP, IAX2 or h323. This DID Phone Number will receive calls using a sip soft phone, hard phone, or an IP PBX. The charges are charged per month, and per min, and per channel based. You do not require a hardware card in case you have a DID service provider to receive call, ie a PRI Card or a Analog Card. The call travels to you all the way over the Internet

Cheapest DID Providers


Algeria

  • Algerian DID numbers Currently only national Algerian numbers.. Also 68 other countries available. Free forwarding to Voip/SIP, IAX,H323 Google Talk, etc many providers preconfigured. Forwarding to PSTN (landlines and mobiles) from 2cent/min, including mobiles. Mobiles Europe 6cent/min.

Argentina

  • Phone2call Virtual Numbers DID are available in 60+ countries including Argentina. Free forwarding to Voip/SIP (some preconfigured providers), IAX, H323, Google Talk, Virtual PBX (yes, free Hosted PBX). Also, very cheap call forwarding to PSTN (regular landlines and mobiles). No contracts. Pay per month, Instantaneous Activation, Many payment methods. Try not only our Virtual Numbers DID but also our free Hosted PBX which has many features (useful for example for Small and Medium Business). Know why we are known for our VOIP leadership.
  • Argentina DID Numbers€ 4.99/month| DomesticNumbers offers Argentina virtual phone numbers from 19 different cities in Argentina, including Buenos Aires. Website also in Spanish.

Australia

  • Phone2call Virtual Numbers DID are available in 60+ countries including Australia. Free forwarding to Voip/SIP (some preconfigured providers), IAX, H323, Google Talk, Virtual PBX (yes, free Hosted PBX). Also, very cheap call forwarding to PSTN (regular landlines and mobiles). No contracts. Pay per month, Instantaneous Activation, Many payment methods. Try not only our Virtual Numbers DID but also our free Hosted PBX which has many features (useful for example for Small and Medium Business). Know why we are known for our VOIP leadership.
  • Comfax Australia Free Fax Free Fax to Email Corporate Services

Austria

  • Sipgate.at Free personal 0720- national number in Austria.
  • TeleCallMart Unlimited Incoming Phone Numbers, Voip calls, SIP Phone, Auto Attendant, DTMF. No monthly fees. Low prices!


Bahrain

  • Phone2call Virtual Numbers DID are available in 60+ countries including Bahrain. Free forwarding to Voip/SIP (some preconfigured providers), IAX, H323, Google Talk, Virtual PBX (yes, free Hosted PBX). Also, very cheap call forwarding to PSTN (regular landlines and mobiles). No contracts. Pay per month, Instantaneous Activation, Many payment methods. Try not only our Virtual Numbers DID but also our free Hosted PBX which has many features (useful for example for Small and Medium Business). Know why we are known for our VOIP leadership. ...

CyberData

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www.CyberData.net

Products:

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Ceiling Speaker V2.0
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Cyberdata Intercom
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Cyberdata Intercom





Cyberdata PoE VoIP Paging Gateway

Cyberdata VOIP Gateway


The CyberData SIP-enabled Paging Gateway allows the use of legacy analog zone paging amplifiers when converting to a VOIP system. The Gateway is compatible with most SIP-based IP/PBX servers that comply with the SIP RFC 3261. The Gateway is powered via PoE 802.3af - no external power supply is needed. ~ description via website


Notes on setup in FreePBX using a SIP Extension:
Set Qualify = no to get it to work.

WAudette...

For more information contact;
Bill Majerczak
billm@cyberdata.net
831-373-2601 x-102

Cyberdata PoE VoIP Speaker


An overhead paging speaker that connects direct to Ethernet.

From the Data Sheet:

The CyberData PoE VoIP Speaker is a
Power-over-Ethernet (PoE) and
Voice-over-IP (VoIP) public address
loudspeaker that easily connects into
existing local area networks with a single
cable connection. The speaker is powered
via a standard ethernet cable – no external
power supply is needed. With its small
footprint and low height, the speaker can be
discreetly mounted almost anywhere.


New in 2009 - VoIP Talk-Back Speaker

^
The CyberData SIP-enabled Talk-Back Speakers is a Power-over-Ethernet (PoE802.3af) and Voice-over-IP (VoIP) public address two-way loudspeaker that is easily connects into existing local area network with a single CAT5 cable connection. The speaker is compatible with most SIP-based IP PBX solutions. In a non-SIP environment, the speaker is capable of broadcasting audio through a multicast. It’s small footprint and low height allows the speaker to be discretely mounted almost anywhere. Speakers can be mounted into existing ceilings or in one of our enclosure kits. Enclosure kits offered are beveled wall mount adapters and NEW in 2009 our wall mount adapter with a digital clock perfect for schools.


See Also



Gentek
Westcon
888VoIPstore
Neobits
Anixter
VoiPmodo.com

Where to Buy



Asterisk embedded systems

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Configuring a tiny Asterisk system


There is a lot of interest in running Asterisk on "embedded" plattforms like the Linksys NSLU2.

News

  • 2013-06-29 Nicherons: ipPBX-mini SpoTel mini ipPBX: supports wireless wifi connection
  • 2013-03-15 Nicherons: ipPBX-mini SpoTel mini ipPBX: Now supports External USB drive/memory stick for call & conference recordings.
  • 2013-03-07 beroNet: OpenPBX beroNet OpenPBX: embedded Asterisk including GSM/FXO/FXS/BRI support
  • 2012-10-15 Nicherons: ipPBX-mini SpoTel mini ipPBX: Big phone system solution in the most elegant & portable device.
  • 2012-04-05 Nicherons: M200P Mobile TDM card, which supports GSM, CDMA, FXO, FXS ports + SMS module. A great hybrid card that supports 5 different module types for building small embedded system.
  • 2012-1-5 Switchvoice: BR4 Appliance - Quad ISDN BRI port PBX based on Blackfin DSP, PR1 Appliance - Single span ISDN PRI PBX based on Blackfin DSP, both using Switchfin - an open source uClinux telephony distribution
  • 2011-12-14 Nicherons: TDM1600P version 2 is a very high-density Half-lenght TDM card for building small compact systems up to 16 analog ports
  • 2011-11-09 Nicherons: ipPBX02 has released the most compact embedded Asterisk ip PBX phone system
  • 2011-08-08 Nicherons: SPoGSM now supports Call Queue, Blacklist, Conference recording
  • 2011-06-20

personal loan

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One particular thing traders like about them is they process personal loan considerably faster. When you utilize to get a loan from classic lenders, it's important to wait for no less than 30 days. If the software is turned down, then that is a squandered month. Within the other hand, difficult money lenders need to have only some times to approve or reject applications.




IP PBX

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IP PBX is a phone system utilising IP communications. Traditionally IP PBX's are located on site where they can also interface to traditional telco services such as analogue phone lines. The business end users connect via IP to the IP PBX for voice service.

Vendor Information


The following is a list of IP PBX vendors, manufacturers, and service providers.
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3CX - 3CX VoIP IP PBX
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Allo NANO2 asterisk pbx for Soho and SMB

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Bullroar Telecom - Virtual PBX service from Bullroar is an all-inclusive hosted VoIP solution serving small businesses across the Gulf Coast. For one low monthly price per phone users have full access to a broad portfolio of features, virtually unlimited nationwide calling, and the reliability that comes along with our distributed nationwide network.
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Diamondcard Hosted PBX - Diamondcard Hosted PBX
Diamondcard Hosted PBX - Diamondcard Hosted PBX Wiki

Lean, mean and easy to use. We do initial configuration for you so its really easy to setup. Low cost and using open Source technology. Get started for as low as $9.60/mth. Reseller and private label program too. +1-303-997-0900 #1.

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.e4 Technologies - . ...

Digium

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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

DID

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Direct Inward Dialing Number (Also known as DID or DDI)


DID (DDI) Background
Most businesses have several incoming telephone numbers used for specific purposes. For example customer service, sales, etc. Some have an individual telephone number for each user in the system. In a home setting on the other hand, each telephone number comes in on a different pair of wires typically. This is not practical in a business enviroment that has many telephone numbers.

Why was DID actually Created?
So DID ("direct inward dialing") was invented as a way to re-use a limited number of physical phone lines to handle calls to different published numbers. In a business with DID, the phone company uses DID signalling to identify the number they are about to connect to the business's PBX. Historically, this was done by pulsing the last 3 or 4 digits of the number being dialed before connecting the number. The PBX would use these DID digits to switch the call to the right recipient.

In modern PBX's, typically, digital methods (example: PRI) are used to do the same thing, ie. supply the "called party" information. But many business's still have old PBX's which use the analog signalling I mentioned before. The type of telephone lines used for analog DID are different than regular home telephone lines. Usually, battery voltage is supplied by the business PBX instead of the telco. Also, the telco signals a new call by bridging the line instead of by ringing the line. The receiving PBX signals back that it's ready to take the call by momentarily reversing polarity of the voltage on the line (this is called "winking" the line)

Old Fashion Way: (PSTN WORLD)
Direct Inward Dialing is used when your PBX telco connection allows direct dialling to extensions within a PBX, using physical lines (or channels on a PRI) on a shared basis. DID service consists of identifying the "called party" by using DTMF or by digital means, before connecting each call. The service can be sent over an E&M Wink T-1 as DTMF and also as D-Channel information on a PRI.

On a PRI connection, the telco can send only the digits that differ between the group number and the extension (often four digits) or the whole number - it depends on the connection to the telco.

DID (DDI) in the new VOIP World

Let's say you buy a phone line from Vonage or some other phone service provider who offers phone service over broadband. The number that they provide to you, in technical terms is a DID number. This is the number that they have assigned to you to connect you to the old PSTN Networks around the world. Any service provider who wants to offer a phone service over IP address, needs to buy DID numbers from his CLEC or any other large service provider like Level 3 in the United States or go to a consortium (company that will take large blocks from many providers and hand them out one at a time)

If you are using an IP PBX like Asterisk, and you want to connect yourself to PSTN so people can call your office, you can either

1) buy an Analog or E1/T1 card from Digium, OpenVox, Rhino Equipment Corp or sangoma

2) buy DID number from DID service provider

DID Service Providers, convert the analog to digital and provide these DID numbers over the Internet, with SIP or IAX2.

Service providers such as DIDnumbers.com, DomesticNumbers, DIDForSale, DIDWW,

Virtual PRI

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Virtual PRI services is an optimum solution for businesses that require quality, high capacity local access numbers without having their own facilities. The Virtual PRI provides immediate turnup and allows expansion into new regions and markets with very minimal cost and investment.

Typical applications for Virtual PRI service are:

  • Call Centers
  • CallingCard Applications
  • Conference Applications
  • High volume incoming calls businesses


Virtual PRI Provider List (In Alphabetical Order):


Alcazar Networks - Over 3,100 rate centers. Virtual PRI as low as $46.00 for 23 channels. DID numbers as low as $0.10/each. Stop buying from the middle man.

CallnFax PRI for business - We offer PRI services in several cities, and the customer support your business needs. We offer hands on service custom tailored to your business needs

DIDLIVE Virtual PRI Offers DIDs + 24 incoming channelsor more in the USA and Canada with over 4000 cities available. Call 1-212-901-0800 or visit www.didlive.com for details.

DIDWW Virtual PRI Offers DID + 30 incoming channels in 23 countries for 50$ a month, revenue sharing model available.

DomesticNumbers Unlimited channels DID + unlimited incoming channels for €19.95/month covering 70+ countries. Also International Free Phone (Toll Free) and Premium numbers available.

MyVNumber Quite cheap Virtual PRI DIDs with 30 channels per DID in 25 countries www.myvnumber.com

Phone2call Offers Virtual PRI with 30 channels (up 30 simultaneous incoming calls) around 25+ countries. You can forward your Virtual PRI to regular mobiles/landlines (PSTN), VOIP, Google Talk, Linphone, Virtual PBX. Only 2 USD/channel. Know why we keep VOIP leadership!

Also See

Enterprise VoIP

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Enterprise VoIP is a term to describe the unified communications systems typically utilized by medium to large businesses and corporations. Most large businesses employ an enterprise VoIP solution instead of a traditional telephone system in order to drastically cut costs and increase their bottom line. In addition, enterprise VoIP providers offer solutions that extend far beyond what is possible with a traditional telephone system.

This is a list of enterprise VoIP providers. These companies typically support multi-line telephone systems, PBX gateways, on-premise PBX systems, and hosted VoIP. Please add enterprise VoIP providers to the list below.

  • 1Pipe Telecom | OnePipe is a CLEC providing SIP Trunking, Hosted PBX, IP and other services, we provide services on our own network and gateways.
  • 45meeting Simple online conference solution for small business.
  • Affinity SIP is a Business VOIP service provider. Great rates for international VOIP calling U.S. DIDs $1.00, Inbound/Outbound $0.009/min, E911 and CNAM free! Wholesale SIP Trunks. Service works with all IP PBX including Asterisk, FreeSWITCH, SIP, IAX2, Elastix, trixbox®, FreePBX, and more. Redundant gateways for 100% up-time. Try SIP Affinity now risk free! Instant activation with free test credit.
  • ActivePBX™ | Turn-Key Business Phone System $149/mo.
  • 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 capability on 100% of our DIDs, CNAM, e911, and local number portability. Wholesale SIP
  • Big Bang Systems provides on premise and hosted pbx and ivr solutions for company with 50+ employees.
  • Bluestone Communications provides enterprise class hosted PBX services with QOS measurement from our servers to your site for a fundamentally better service. Extensive array of facilities and apps, just $15 per phone per month.
  • Net-Voice Telecom NVT is a Leader in IPPBX and Sip Trunking with Office Located in NY and FL we have extremely competitive Packages for Small to Enterprise class Businesses
  • VOIP Essential offers high quality SIP termination services to businesses in the US and Canada. We specialize in Dialer Traffic and provide amazing flat rate plans to low/high volume call centers and enterprises. We also provide wholesale services to several destinations including the Americas, Europe, and Asia. Call now for instant account setup and free test credit. Contact: Sean on +1(631)2292925 or visit www.voipessential. ...

GSM

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GSM stands for Global System for Mobile Communications. It is the dominant mobile phone comminucation protocol in Europe and also popular elsewere. The name is also used to describe some of the voice codecs used in the GSM standards.

This page regards connecting an IP-based phone system directly to the GSM cellular network in addition to connecting a VoIP PBX to the GSM network & PSTN; all-in-one-box. For more information about the codec, see the GSM codec page.

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VOIP GSM Gateways

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What's a VoIP GSM Gateway?

A VoIP GSM Gateway enables direct routing between IP, digital, analog and GSM networks. With these devices (fixed cellular terminals) companies can significantly reduce the money they spend on telephony, gp-especially the money they spend on calls from IP to GSM. The core idea behind cost saving with VoIP GSM Gateways is Least Cost Routing (LCR).

Through least cost routing the gateways select the most cost-effective telephone connection. They check the number which is dialed as well as rate information which is stored in an internal routing table. Because several SIM cards and GSM modules are integrated within the VOIP GSM Gateway it is able to make relatively cheaper GSM to GSM calls instead of expensive IP to GSM calls.

Who offers VoIP GSM Gateways?


HyperMedia Systems

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Hypermedia Systems


Hypermedia Systems develops and manufactures scalable, professional GSM Gateways and VoIP GSM Gateways for low to high volumes of traffic, starting from 8 ports, with up to 72 cellular channels/ 288 SIMs. Available interfaces: VoIP, E1/T1-PRI and Analog interfaces, as well as GSM, 3G UMTS, CDMA.

These carrier grade Voice Gateways provide opportunities for Alternative carriers, Call termination vendors, and Service providers, as well as for SMB and corporate enterprises.

Read our Testimonials page.


Visit Hypermedia's Website

We encourage you to visit and get to know it. Find many interesting new areas, like a new downloads page where you can find White Papers, Data Sheets, and Application Notes and more. See our latest addition: Clips & Presentaions.

Update:Meet with us at WWC 2013 - Check our new Termination Systems!


GSM Gateways and VoIP Gateways for Call Termination

Compact bi-directional gateways for providing cost-effective termination services to or from GSM, UMTS/3G, CDMA, VoIP, E1/T1-PRI or Analog.

Special Offer till 30/06/2013 - Safe revenues with heavy duty termination system! For more information press here.
      • Applies to HG4032p and HG4016p heavy duty series only
      • SLA required


  • Fast Start Series for Call Termination - Plug 'n Sell VoIP GSM Gateway. An ALL-IN-ONE solution for the termination business with a minimum investment.
  • Professional VoIP GSM Gateway. HG-4000 Series. From 12 to 72 cellular ports.
  • Call Termination Systems for VoIP/3G, GSM & CDMA.
  • Combination UMTS / GSM Gateway - HG-5000 Series. The right opportunity for anyone who plans to provide termination services by using both GSM and UMTS cellular networks. You can have several combinations of technologies in the same unit, and receive the traffic by using Embedded VoIP Gateway or by E1/T1-PRI interfaces.


HG4000_side.jpg

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

VirtualPBX

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XVB - Voice Application Server / HostedPBX solution / HostedIVR solution based on asterisk ( v.3, build 7490 ).


The virtual-pbx application is designed for processing incoming/outgoing calls in an isolated environment (numbered plan, routing calls, phones, cdrs, web gui and so forth. ) for multiple users.

Virtual-pbx application include :

  • Processing incoming / outgoing calls.

  • Completely isolated environment for different users ( incoming / outgoing routes, dial plan, sip-endpoints, web-interface, cdrs, call-recordings, etc ).

  • Full control of users ( disk quota, the number of phones, the number of messages, the number of concurrent calls, etc ).

  • Custom user greetings support.

  • Email/Twitter notifications.

  • Multiple language voice prompts.

  • Text To Speech ( TTS ) for multiple languages.

  • 'Simple' or 'Expert' mode for configuration.

  • Custom music on hold (MOH) for each user.

  • Multiple language WEB interface

  • xml/json API support.

  • Flexible customisation for system voice messages.

  • Managing voice mail via phone or WEB interface.

  • User specified time zones support.

  • Call transfer support ( blind transfer, with return of unsuccessful call ).

  • Smart call pickup.

  • White / Black lists support for each IVR item.

  • PhoneBook with speed dial feature.

  • Journal configuration changes.

  • CDR support ( search by caller, called, date of call, etc.. )

  • XML backup / restore configuration.

  • Multiple roles within a single IVR account.

  • Privated / Shared DIDs support.

  • SQL reports.

  • Management API.

  • Call routing based on Google.Calendar events.

  • Support Google Analytics for calls trekking.

  • Radius accounting.

  • Background music for Find-Me / Queue calls.

  • Asterisk 1.8 ( optional ).

  • Support for wideband / HD audio codec ( g722 ).

  • Email/Web interface Branding.

  • Recording outgoing calls ( auto / on demand ).

  • Full DTMF history for each call.

  • Support presettings for SIP providers.

  • Support Multi-tenant asterisk with Kamailio as sip registrar server / load balancer.

  • Easy to update.

  • speedyAGI server ( greater stability && high performance )

  • FMC - Fixed Mobile Convergence / One touch call transfer without breaking the call.

  • Google-chart for extended statistics for: DialOut, Queue, CallBlast, Polling, etc

  • WEB logon / logoff for the agents of the queue. ...
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