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Virtual PBX

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Virtual PBX is a budget-friendly form of hosted VoIP (Voice over Internet Protocol) that usually only handles inbound calls. A virtual PBX is typically intended for small business VoIP customers with fewer than 10 employees and low-volume telephone traffic.


What Is Virtual PBX?

A virtual PBX is an economy-class version of hosted PBX. Hosted and virtual PBX systems are business VoIP PBX phone systems that transmit calls over the Internet as data.

A virtual PBX offers inexpensive business VoIP telephone service to small businesses. As with a hosted PBX phone systems, a virtual PBX is owned and maintained off-site by a VoIP service provider. A virtual PBX enables a small business telephone system to access enterprise-level features such as auto attendants and voicemail. With virtual PBX small business telephone systems, small start-ups, mom-and-pop shops, freelancers, and entrepreneurs can all present a professional image to vendors, investors, and customers.

Depending on the service provider, a virtual PBX phone system may require a separate phone service for outbound calls.

Virtual PBX Features


Virtual PBX phone systems offer lower costs and fewer features than hosted PBX phone services. Compared to hosted PBX small business telephone systems, virtual PBX service is limited to the most basic fundamentals of business-oriented call controls. Virtual PBX is geared toward simple inbound call-routing for SoHo offices with few personnel, small budgets, and limited calling needs. As with many hosted PBX calling services, most virtual PBX phone systems do not require a contract or term commitment.

Standard features offered with most virtual PBX plans are:

  • Voicemail
  • Auto attendant
  • Unlimited call handling (no busy signals)
  • Call forwarding

Limitations


Virtual PBX phone systems generally:

  • Handle only inbound calls
  • Offer a limited number of extensions
  • May not include Fax over IP (FoIP) services
  • Include a set amount of free minutes
  • May not offer voicemail-to-email
  • May not include international long-distance coverage
  • May not offer Internet fax service
  • May charge extra for conference calling

The features offered vary by virtual PBX VoIP provider. As VoIP service becomes a more common solution for small business telephone systems, many virtual PBX plan features are incorporating the more advanced features of hosted PBX phone systems. Compare plans and prices to determine the best virtual PBX solution.

Cost

Virtual PBX phone prices depend on a variety of factors, such as the features included. Virtual PBX phone service plans can start as low as $9.95 (Grasshopper) per month.

Virtual PBX Service Providers


Some virtual PBX providers include:


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?


Asterisk Consultants Italy

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Web Hosting

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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

Automatic Call Distributor

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Automatic Call Distributors

Automatic Call Distribution or ACD, is a tool commonly used in the telephony industry. ACD systems are commonly found in any office that handles a large volume of inbound calls. The primary purpose of an Automatic Call Distributor is to disperse incoming calls to contact center agents or employees with specific skills.

The ACD system utilizes a rule based routing strategy, based on a set of instructions that dictates how inbound calls are handled and directed. These rules are often simply based on guiding a caller to any agent as fast as possible, but commonly multiple variables are added, all with the end goal of finding out why the customer is calling. Matching and routing literally thousands of calls to the correct agent is a difficult task, and is often done in concert with Interactive Voice Response and Computer Telephony systems. ACD servers can cost anywhere between a few thousand dollars to close to millions of dollars for a very large call center handling thousands of calls per day.

Automatic Call Distributor Vendors

  • VadaXchange for Call Centres & Contact Centres In addition to Asterisk based ACD, VadaXchange Buddy provides full call centre agent and manager experience. It allows call centre agents to easily log in and out of the queues and see who else is in the queue. It also allows managers to monitor the performance of their agents on-line in real time, wherever they may be situated.
  • 3CLogic offers a full suite of cloud based inbound and outbound customer interaction channels such as voice, chat and social media which integrate with our clients' existing CRM software such as Salesforce.com and Microsoft Dynamics. This integration provides businesses with a 360 degree view of all customer interactions regardless of the channel chosen by their customers.
  • Convox: ACD Solution by Deepija Telecom Pvt. Ltd. Feature- Auto Dialing, IVRS,CTI Integration, Vtiger Integration, Voice Broadcasting, Lead management.
  • UCN has established itself as a leader in the telephony industry by providing hosted telephony services, that match their traditional counter parts in quality, but that can be attained at a fraction of the cost. UCN has a myriad of contact center tools, including IVR, Automatic Call Distributors, & Computer Telephony solutions. If you're looking for a comprehensive solution, you may want to consider UCN's Hosted Contact Center solution inContact, which includes Hosted ACD Software by UCN, Computer Telephony Integration, Hosted IVR Software, Workforce Management Software& numerous other Hosted Call Center solutions.
  • Database Systems Corp.ACD - Integrated

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.

IP PBX

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IP PBX is a service where the call platform and PBX features are hosted at the service provider location. The business end users connect via IP to the provider for voice service.

Vendor Information


3CX - 3CX VoIP IP PBX
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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 - .e4 provides powerful, integrated IP Communications enterprise class services that provide the SMB customer to configure the exact voice, video, messaging and collaboration services they need in just minutes- all delivered on-demand.
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Ip-Pabx.comIP-Pabx.com - First of its kind Hosted IP Based Pbx system offering from A-Z Solution from one provider. Totally Hosted Web Based Pbx System, Very Low priced starting 199$. You place an order, choose your Country Code and Area code for your incoming lines, choose extension quantity, They ship you pre programmed 99$ ip phones, you plug them on the internet, and wala, you have a pbx system up and running. the extensions can be anywhere in the world, and they will work fine. visit the web site for demo, or email us for a free test account. You get Voice Mail, Fax to Email, Web Voice Mail, Call Recording, 20 IVR's. API to integrate your own applications. ...

Asterisk Paid Support

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Asterisk AGI php

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Using PHP with the Asterisk Gateway Interface (AGI)

PHP Tips and Examples

Script Location

Asterisk expects to find scripts in the /var/lib/asterisk/agi-bin/ directory for AGI execution.

File Permissions

Remember to make your files executable with chmod 755 /var/lib/asterisk/agi-bin/*.php .

The Shebang

The first line in your script will of course determine which executable runs it. Be aware of the different versions of PHP, such as /usr/bin/php and /usr/bin/php-cgi. In most cases life will be easier if you use the CLI version— typically at /usr/bin/php. (Use /usr/bin/php -v to confirm that you are using the CLI version.) This will make $argv and $argc available, disable buffering, disable HTTP header output, and pipe error messages to STDERR instead of STDOUT. All these are desirable for working with Asterisk. If you use the CGI executable, you will have to use some command line options and edit your scripts to ensure proper behaviour.
The shebang should be followed immediately by the <?php script opener, with only a newline between them. Any whitespace could get sent to Asterisk, causing problems.

Read AGI Inputs

All communication with the AGI occurs over STDIN and STDOUT; these IO streams are predefined in PHP and do not need to be opened or closed. When your program starts, Asterisk will stream a number of variables to it, which it's in your interest to capture. ...

VoIP Providers USA

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This is a list of VoIP providers in the USA. These companies typically support multi-line telephone systems, small PBX gateways and hosted VoIP. Please add VoIP Providers USA to the list below.

  • 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.
  • Cheap VOIP Inc Wholesale to the public, great rates and good quality.
  • DIDforSale Buy unmetered Inbound SIP DIDs for just $7.50 all over us. All DID comes with 20 channels.
  • DLS Hosted PBX offers the most call center and integration features
  • CheapVOIP4U SIP termination starting at $0.0027 per minute without fees. Pay as you go service.
  • CircuitID provides high-quality Wholesale VoIP Termination, Origination, Toll-Free, and Microsoft Lync SIP Trunking services, at a low affordable cost. Experience carrier class quality, and savings benefits.
  • Flowroute LLC Wholesale VoIP, A-Z SIP Termination, Cheap DIDs, Free CNAM Storage, E911, T. ...

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

see Cheapest ATAs and Service
see domestic USA DIDs for carriers

Free Service Providers Only (Free DIDs)


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 Virtual Numbers $5.95 Per month / Free Setup| Virtual Numbers in Argentina with Free Forwarding Options(FlyNumber)
  • 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


Austria

Secure VOIP Service Providers

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Marketing is NOT ALLOWED on this page. Please describe services in neutral language. If you want to add your company, please read the Posting Guidelines for Promoting Products and Services. Do not use UPPER CASE, do not use bold face, do not use italics, or exclamation points. Do not use the word "best". Please indicate whether you are a facilities-based provider or a reseller. Use neutral language and play fair. Please place your entry in alphabetical order with respect to other vendors in the same section. And if you make your own category, it will be deleted.


North America

  • CI Gear CI Gear Business VoIP services include Hosted Phone Systems and Business VoIP Lines delivered securely over the Internet to your VoIP Phone, VoIP Adapter, VoIP Phone System or VoIP Gateway.



Middle East & Asia Pacific



South & Central America



Europe




New Software Releases

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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.

November 2012


October 2012

Asterisk High Availability Solutions

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Ways to increase system availability and balancing:

  • HAAST from GenerationD www.GenerationD.com offers high availability / clustering for your existing Asterisk servers. The High Availability ASTerisk (HAAST) add-on offers rapid automatic failover of a failed peer, transparent to up/downstream devices. HAAST also supports manual promote/demote for maintenance, a command line interface, a telnet interface, a web based interface, and IP address sharing for simple configuration and operations. Installation is straight forward, with no additional hardware required. Real-time fail-over takes 3-10 seconds to complete. Turn any pair of Asterisk servers into a cluster. HAAST is a commercial solution, in use at call centers and other high-uptime environments. A "Lite" version is also available for smaller Asterisk installations.

  • HiPBX is a new FOSS highly-available enterprise ready CentOS6/Asterisk/FreePBX distribution. It is currently being written by one of the original developers of FreePBX, and tightly integrates with (currently) Cisco SPA model phones. It provides Active Directory integration, automatic phone provisioning, user-configuration of phone features through their own Web interface, and cluster and node management through a GUI. This is all new software, and is not based off an existing project. It's currently under intensive development, and all source code is available on Github.

  • DNS SRV on the CPE side but not all phones handle this.

  • SARK-HA from Aelintra Telecom offers High Availability Asterisk out-of-the box. Runs Aelintra's SARK UCS MVP on a pair of servers.... Real-time failover takes less than 20 seconds to complete and includes support for ISDN PRI circuits. The servers are kept in synch using rsync. Wiki pages HERE. System also includes multi-tenant and a fully integrated provisioning system with zero touch, DHCP-free set-up for multicast capable phones... see HERE.

  • Ranch Networks offers High Availability White_Paper_one_one_HA.pdf solution for Asterisk. This is Hardware based solution. (Just for two asterisks boxes).

  • Flip1405 Manages virtual IP between two Asterisk servers and queries UDP5060 for state changes
    • Downtime less than 30 seconds
    • Only 2 dependencies (nmap and arping)
    • Incredibly easy to setup

  • SERVERware is a next generation communication technology solution. Delivers a wide range of IP services and applications. It is redundant, high available and fault tolerant. The SERVER edition of SERVERware is the most economical way to start delivering IP service from a single server. This edition includes all the necessary components that allow service providers to offer any of the supported VPS templates with a clear path upgrade to network edition.

Asterisk consultants Asia

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This is a page about Asterisk consultants in Asia. Please keep this list alphabetical.

Asia

BANGLADESH

E-NET COMMUNICATION

We provide the following consultancy:

    • Asterisk SS7/E1/T1 Solution
    • Asterisk GUI
    • Asterisk AGI programing
    • Asterisk Based Call-Center
    • Asterisk Calling card/callback
    • Asterisk IPPBX
    • OpenSIPS/kamailio/Yate/GnuGK Consultancy
    • Asteisk Billing
    • Asterisk at large environment

Please contact:
    • Contact: Mohammad Emran
    • Phone: +880-1713033045, +880-1768200000
    • Address: HB Tower(3rd Floor),House No# 1/A,Road # 23, Gulshan-1,Dhaka-1212,Bangladesh.
    • Email:mailto:monemran@gmail.com


CHINA/SINGAPORE/MALAYSIA/HONG KONG/TAWAN

BangJian telcom Ltd.

BangJian telcom Ltd located in ShenZhen, China. BangJian Telcom provides PBX, CRM, Call center, SS7/PRI solutions and asterisk cards. Lists:

CHINA


asterCC technology Co., Ltd.

asterCC is a software development company, our main product is asterisk based IP PBX & Call Center system.
We have been active in software development for computer telephony and Internet, for over 5 years.


    • asterCRM open source call center & crm
    • asterBilling hosted call shop & realtime billing solution.
    • asterCC Commercial hosted PBX & Call Center solution, provides free ip pbx features and 5 free agents license.

WShuttle Infotech Ltd. ...


Session Border Controller

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A Session Border Controller is a device used in select VoIP networks to exert control over the signaling and usually also the media streams involved in setting up, conducting, and tearing down calls. The SBC enforces security, quality of service and admission control mechanism over the VoIP sessions.

The Session Border Controller is often installed in a point of demarcation between one part of a network and another. Most Session Border controllers will be installed between peering service provider networks, between the enterprise network and the service provider network, or between the service provider network and residential users.

A Session Border Controller is like a Firewall for VOIP.
They are often configured as a SIP Back-To-Back User Agent (See SIP RFC).

In addition to firewall functions they also may provide services like NAT traversal.

See: NAT and VOIP for more information and watch the video on The Anatomy of Session Border Controllers and To Couple or Decouple Routing Intelligence from SBC.

Martyn Davies of Dialogic discusses Session Border Controllers as the often misunderstood "black magic" application for VoIP networks.


Although carriers often use Session Border Controllers for signal translation and security, most do not include the hardware-based signal processing needed for media transcoding. For all-IP environments, new elements are required that can mediate signaling, transcode among different media formats, and handle basic security issues. The concept of Multimedia Border Element (MMBE) meets these needs.

Products




Where to Buy


Twitter VoIPUser Directory

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This is a page that can contain entries of Twitter users and their business or VoIP interests.

Please insert into correct alpha place and please leave things clean enough to be helpful to people on Twitter who want to find VoIP Users

Voip-info.org Twitter: @Voip_info


OverSIP

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the SIP framework you dreamed about

www.oversip.net

Overview


OverSIP is a SIP proxy & server built on top of Ruby EventMachine-LE async library which follows the Reactor Design Pattern.


Features


  • Fully asynchronous event-based design.
  • SIP over UDP, TCP, TLS and WebSocket transports.
  • Full support for IPv4, IPv6 and DNS resolution (NAPTR, SRV, A, AAAA).
  • Fully programmable using Ruby language.
  • The perfect Outbound Edge Proxy.


Resources




License


OverSIP is available under the MIT License.


See Also




JsSIP

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''the JavaScript SIP library"

jssip.net

Overview


JsSIP is a pure JavaScript SIP stack using the SIP WebSocket Transport for signaling and WebRTC for audio/video.

Features


  • SIP over WebSocket (use real SIP in your web apps).
  • Audio/video calls (WebRTC), instant messaging and presence.
  • Lightweight! (~140KB).
  • 100% pure JavaScript built from the ground up.
  • Easy to use and powerful user API.
  • Works with OverSIP, Kamailio and Asterisk servers (more info).
  • Written by the authors of draft-ietf-sipcore-sip-websocket and OverSIP.


Resources




License


JsSIP is available under the MIT License.


See Also



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

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