Unicoi Systems, Inc.

November 13, 2012, 3:59 pm

Website: www.unicoi.com

About Unicoi Systems

Unicoi Systems is a leading provider of real-time voice, video and data software and reference designs to embedded device developers and OEMs worldwide. Founded in 2000 in Atlanta, Georgia, over 500 leading companies rely on Unicoi's Fusion Embedded™ Software to power their solutions.

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Hosted PBX

November 14, 2012, 4:13 am

≫ Next: Sip Trunking Providers

≪ Previous: Unicoi Systems, Inc.

Hosted 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.

"Hosted" means to say that the hardware and PBX is hosted at an off-site location from where the VoIP telephone service is being used. An office can have VoIP telephone service that powers their phones in the office, but their PBX could be hosted at their VoIP providers data center, thus the term: hosted PBX. Hosted PBX is also sometimes referred to as Hosted VoIP.

Benefits of Hosted PBX

There are many benefits to using Hosted PBX rather than a traditional phone system, or an on-premise PBX. The main benefit is cost- a Hosted PBX system costs much less to set-up than an on-premise PBX. In many cases, there are no set-up fees for a hosted PBX system. Purchasing and setting up an on-premise PBX can cost tens of thousands of dollars. Hosted PBX phone systems fall under operational expenditure rather than capital expenditure, which also makes hosted PBX service attractive to businesses. With hosted PBX service, you pay a monthly fee, and the hosted PBX service provider takes care of the rest.

Another benefit to a hosted PBX system over an on-premise PBX is that hosted PBX service providers will take care of all the set-up and installation, meaning you do not need to be a telecom or VoIP expert in order to get a hosted PBX system. A downside to a hosted PBX is you may have a little less of an ability to customize your solution to your business, but many hosted PBX service providers can achieve a deep level of customization.

More Hosted PBX Service Providers

1Pipe Telecom | OnePipe is a CLEC providing SIP Trunking, Hosted PBX, IP and other services, we provide services on our own network and gateways.

Actual US CLEC
Branded customer portals
Multiple locations on one virtual PBX
Customized pricing.

AstraQom Canada | AstraQom Canada is an innovative provider of Hosted PBX, VoIP IP Telephony, Hosting, and Web Services for Business.

All Packages Customizable
White label/turnkey packages available
Virtual Numbers, Vanity Numbers and Toll Free Numbers
AstraQom can provide unified communications for nearly all levels of business

.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.

1 MONTH FREE TRIAL
VoIP and PSTN Service
PBX and ACD
Web, Audio & Video Conferencing
Skills Based Routing
Private/Encrypted and Public IM
Voice and Data Archiving
Reporting and Archiving
PSTN Gateway
CRM Integration
Web Contact Center

1aVOIP.com

1aVOIP.com delivers hosted PBX services with SIP& Mail feature sets. ...

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Sip Trunking Providers

November 14, 2012, 4:39 am

≫ Next: Voip.com Cloud PBX for Business Reviews

≪ Previous: Hosted PBX

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

Country specific pages:

1comms VoIP provider for UK Businesses. Sip trunk services, UK DDI's and Asterisk Support for business users and resellers. White Label fully itemised per second billing.

1Pipe Telecom | OnePipe is a CLEC providing SIP Trunking, Hosted PBX, IP and other services, we provide services on our own network and gateways.

Alcazar Networks - Wholesale Services Over 3,100 DID rate centers. Per minute pricing as low as $0.0005/minute. Per channel pricing as low as $2.00/channel. DIDs as low as $0.10/each. A-Z termination. Over 1,200,000 numbers in DID inventory.

Amivox free your phone - Lower your communication cost VoIP provider for both consumers and businesses. Offer's free SIP account. Prepaid and very good rates for network termination with premium quality ( Amivox-Out) . Support for iPhone, Android and Blackberry. Shared balance for multiple users. Calling Amivox to Amivox is free - Sign up for free and try out the service.

Anveo offers phone numbers from over 48 countries with instant activation. Anveo's Voice 2.0 Communication and Collaboration Suite with powerful Visual Call Flow technology allows you to visually configure call handling and call termination options for your phone number. Anveo provides FREE SIP trunking and it is one of many termination options available.

BellVoz offers International and Domestic Long Distance Services with VoIP technology, helping business and consumers to reduce monthly telephony expenses.

Best VoIP USA BestVoIPUSA.com offers SIP trunking to private and commercial operators of Asterisk PBX switches. BestVoIPUSA.com also provides all of the well known features of SIP even if you are connecting directly to SIP compliant devices, handsets or servers.

Box Internet Services offers SIP trunking to private and commercial operators of Asterisk PBX switches. Boxis.net also provides all of the well known features of SIP even if you are connecting directly to SIP compliant devices or servers.

Brisnorth Communications Australia Brisnorth.com.au provides SIP trunks, VoIP and SIP Server hardware to Businesses Australia-wide. Carrier-Grade reliable SIP/VoIP services at very cost-effective rates. We can work with your current hardware/phones or upgrade you. We have Plans to suit all budgets and sizes of Business. Contracts and Bundles are optional (Customers are free to go un-contacted and un-bundled) email sales@brisnorth.com.au or call 07 3623 0800

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Voip.com Cloud PBX for Business Reviews

November 14, 2012, 3:16 pm

≫ Next: Asterisk Professionals

≪ Previous: Sip Trunking Providers

Voip.com is a business Hosted VoIP service provider. Read Voip.com business Cloud PBX reviews below or write your own.

Write a Review

Asterisk Professionals

November 14, 2012, 5:33 pm

≫ Next: Open Source Billing Systems

≪ Previous: Voip.com Cloud PBX for Business Reviews

The list of Certified Asterisk professionals around the globe.

Australia

Darley Stephen

MOBILE 0401597387
E-mail: darsdsd@gmail.com
all Asterisk / digium installation & support, small / Large scale deployments, Asterisk configuration, Multiple location deployments
Predictive Dialer
Call Center deployment
Industry based telephony installation
Global PBX setup
IVR
Multi site Maintenance
E1/PRI/ISDN setup
red5 installation & config
flash phone customization

Asia (ASEAN) / Myanmar

Asterisk/FreePBX consultant
Asterisk based IP PBX professional service

The first Asterisk Consultancy in Myanmar ***

1. IVR
2. FXO/FXS
3. Dial plan
4. Conference Phone configuration
5. Voice Mail.
6. Android/iphone/Nokia Phones with native SIP support for Asterisk using wifi PBX
7. Zentyal Linux Server Setup.

Technical Support for products: Avaya Nortel, Grandstream, ATCOM, Openvox, Digium, Xorcom, Sangoma, Cisco

Contact: Nay Myo
Email: naymyowin@hotmail.com

Asia / India

Enterux

Unified Communication talk to Enterprise Oracle Apps!!!

Predictive Dialer
Medium to Large Multi-Office Business Telephony Systems
Call Centers - Local and International
Asterisk Dialers and Bulk Calling Systems
International Office Telephony Systems
Specialty Calling Systems (Entertainment and Personals)
Healthcare Application Integration
Customer Relationship Management, CRM Application Integration
Distributed Server Architecture and Asterisk Load Balancing
SIP Express Router, SER Load Balancing
Hospitality Telephony Systems (Hotel PBX Integration)
Complete IVR Development
Local or Datacenter PBX Customization
Wireless Telephony Installations
Database Integration and Customization
Custom Application Development
Installation / maintenance / configuration of linux systems / servers VOIP Gatekeepers / Phones / devices.
Support for digium / sangoma / rhino E1 / PRI / FXO Digital / analog Telephony Cards
Installation / maintenance / configuration of linux systems / servers MP124 VOIP GatWays / Phones / devices.
Configure Asterisk with Analog phones , USing MP124 Gate Ways or PAP2T Devices
Now make your SAP / Custom Oracle based app talk to Asterisk only available with Enterux,

#1 Asterisk Development and Deployment partner from India.

Connect with us and find out how we can revolutionize the way you communicate

Asia / IRAN

Omid Mohajerani

Digium Certified Asterisk Professional in IRAN
Training Asteirsk and Nagios in IRAN
Website: http://www.miniatel.com

Haamed Kouhfallah

ELastix Engineer and trainer
Web site: http://www.voip-iran. ...

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Open Source Billing Systems

November 14, 2012, 11:46 pm

≫ Next: WombatDialer

≪ Previous: Asterisk Professionals

Billing Systems

This page lists Open Source Billing systems.

Asterisk2Billing (A2Billling):

http://www.star2billing.com/
License: AGPL

A2Billing, combined with Asterisk is a physical Telecom Platform and Soft-Switch providing a wide range of telecoms services using both traditional telephone technology or VoIP. It contains a real time billing engine which rates and bills and invoices calls, and supports payment gateways.

This now gives any Telecom company a very good reason to consider the A2Billing Platform over the traditional offerings for TDM and VoIP Soft-Switches as well as wholesale billing.

Solutions

A2Billing can be used in a number of different roles, and our consultancy services can advise on the appropriate hardware and configuration of your Switch.

Calling Card service with either PIN or Caller ID recognition
Call-back service
VoIP Billing
VoIP termination for IP PBX systems
Wholesale VoIP Termination and Origination
Residential VoIP Termination and Origination.
Special Applications Platforms
Predictive Dialler and Sales Campaign Tool
Hosted PBX, IP Centrex and Multi-tenant systems.
VoIP reseller white label solutions.
and much more :)

There is online demo available at http://demo.asterisk2billing.org/a2b/. For more information about any of the features and benfits of any of our products, please contact us at sales@star2billing.com. For A2Billing installation services, see http://www.star2billing.com/consultancy/managed-install/

Hotelpbx ultimate hospitality tool:

Web Page: https://sourceforge.net/apps/wordpress/hotelpbx/
Autor website: http://translate.google.fr/translate?sl=fr&tl=en&u=http://comdif.com/ Comdif Telecom

License: GPL
English - Spanish - French
This is opensource Cyberhotel version, including great features for hotel.
- This is a live iso it can boot from USB flash or CD or be installed.
- This a complet Debian Squeeze workstation including XFCE interface
- This is not only a PBX it include as whell a captive portal and register users logs in proxy.
This version can work both as standalone GUI or with FreePBX, just change option in configuartion

Gui is pur realtime display, billing is build on a switch with advanced routing features as LCR, SIP carrier backup or on gateway.
This is not an experimental version, but tested long time in production in big Hotels and can have our full support.
Freeware in 2012

Enjoy !

sip:provider CE:

http://www.sipwise.com/products/spce/
License: GPL3+

sip:provider CE is an open-source turn-key SIP platform for up to 50k subscribers. It comes with a powerful and flexible billing and rating engine for near-realtime post-paid billing. ...

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WombatDialer

November 15, 2012, 12:27 am

≫ Next: DID Service Providers

≪ Previous: Open Source Billing Systems

WombatDialer is a platform to provide mass outbound calling on the Asterisk PBX.

This can be used to implement many different services. By offering you a set of ready-to-use components and a monitoring GUI, you can create complex solution in minutes.

WombatDialer can work on pre-defined call lists or can dynamically create them over an API (e.g., dial number X after 10:30 AM). It shares the load on one or more PBX servers and has a flexible rescheduling logic to handle missed calls. It is built to be used with your existing Asterisk PBX and does not require separate servers or a separate set of lines. It can call over VoIP or through the public telephone network. It can also act as a power-dialer for a set of agents logged in in Asterisk.

WombatDialer is built to integrate with your business processes, and can receive calls to be made over HTTP and/or notify an external system in real-time of calls made and results gathered.

WombatDialer works natively with the QueueMetrics Call-Center Monitoring Suite in order to produce state-of-the-art campaign analyses and insight.

WombatDialer is currently available as a free beta version. The current version is 0.6.0 as of November 2012.

See also:

WombatDialer Website
User manuals
Installation instructions
WombatDialer development blog with many real-life examples

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DID Service Providers

November 15, 2012, 7:08 am

≫ Next: Asterisk Queue Callback

≪ Previous: WombatDialer

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

Australia Virtual Numbers $3.95 Per Month / Free Setup | Virtual Numbers in Australia with Free Forwarding Options(FlyNumber)
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!
Austria Virtual Numbers $3. ...

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Asterisk Queue Callback

November 15, 2012, 11:29 pm

≫ Next: Easy To install Java Based Queue Callback

≪ Previous: DID Service Providers

Asterisk Queue Callback

This feature allows a caller holding in your queue to press '1' and enter a phone number to be called back at when their slot in line comes up next. Note: This requires Asterisk 1.2

To accomplish this, I created a SQL table that contains a list of all callers in the queue. When the caller gets put in the queue, an entry is added to the SQL table, and when they hang up it is removed. However, as you can see from the dialplan below, if they press '1' while waiting in the queue, it prompts them to enter in a phone number and then hangs up without removing the entry from the SQL table. I have written a perl daemon that runs in the background and periodically looks at the caller holding the longest in the queue and whether or not they have requested a callback. If they have, the daemon drops a .call file and deletes the entry from the table. The .call file calls the customer back at their predetermined phone number and puts them back in the queue after first setting QUEUE_PRIO to 10 (default callers are set to 0) which puts the caller at the front of the line for the next available rep.

It works well in our small setup, but I'm not sure on how well it would scale. Either way, I couldn't find any answers to accomplish this other than the broken ICD or a commercial offering that was prohibitively expensive.

Anyhow... on to it.. Create the SQL table shown below, and in my example dialplan, you would send calls to the 'support-queue' context. Callers that opt for a callback would get dropped into the 'callback' context (where their QUEUE_PRIO gets raised to 10). Run the perl daemon on your asterisk machine so it can monitor the SQL table and drop the .call files as needed.

Note: This is just for starters. MAKE SURE you add some logic into it to prevent malicious callers from requesting callbacks to 911, 411, the local police, my house, etc.

-- tf. <tyler_AT_unixgod_DOT_net>

SQL Table Structure

If using the dialplan below, put the following table into a database called 'acd'.

CREATE TABLE `callers` (
`uniqueid` varchar(15) NOT NULL default '',
`callback` int(3) NOT NULL default '0',
`callbacknum` varchar(15) NOT NULL default '',
PRIMARY KEY (`uniqueid`),
KEY `callback` (`callback`)
) ENGINE=InnoDB DEFAULT CHARSET=latin1;

Dialplan Additions

[support-queue]
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,MYSQL(Connect connid 127.0.0.1 acd acdpass acd)
exten => s,n,MYSQL(Query r ${connid} INSERT\ INTO\ callers\ set\ uniqueid=${UNIQUEID})
exten => s,n,MYSQL(Disconnect ${connid})
exten => s,n,Queue(support|t)

exten => 1,1,Read(CALLBACKNUM|beep|10) ; This is where you request the CallBack Number from your customer. Put your own prompts or whatever here.
exten => 1,n,MYSQL(Connect connid 127.0.0.1 acd acdpass acd)
exten => 1,n,MYSQL(Query r ${connid} UPDATE\ callers\ SET\ callback=1\,callbacknum=${CALLBACKNUM} WHERE\ uniqueid\=${UNIQUEID})
exten => 1,n,MYSQL(Disconnect ${connid})
exten => 1,n,Playback(goodbye)
exten => 1,n,Hangup

exten => h,1,MYSQL(Connect connid 127.0.0. ...

↧

Easy To install Java Based Queue Callback

November 15, 2012, 11:30 pm

≫ Next: voip-info.org

≪ Previous: Asterisk Queue Callback

Java Based Queue Callback

First off, this is my very first time creating a wiki page. So if I am blowing by some rules here or screwing something up I apologize in advance. Just try to help me out.

So a while back I needed a program that would ask a caller in an asterisk/trixbox queue for there phone number and call them back. I wanted it to be very reliable with nice features. Working for a goverment agency I was able to get ahold of some grant money and I hired a pretty smart programer to develop this code with the intention of share it with everyone. I am hoping others can contibute and keep making this better.

Anyway, attached is a zip file of all the code with a word doc file of the instructions to install.

I would appreciate any feedback and if you do use it, please leave a comment letting me know what you think.

Thanks!

Here is the code:

See also:
- Powerful automated queue call-back for Elastix with WombatDialer

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voip-info.org

November 16, 2012, 6:01 am

≫ Next: Old News

≪ Previous: Easy To install Java Based Queue Callback

Welcome to the VOIP Wiki - a reference guide to all things VOIP.

This Wiki covers everything related to VOIP, software, hardware, VoIP service providers, reviews, configurations, standards, tips and tricks and everything else related to voice over IP networks, IP telephony and Internet Telephony.

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

Update: We have added theFacebook Like and Google +1 button to the top right corner of all pages on Voip-Info.org. Please help recommend the wiki by clicking on them. We also now have Google+ page here and a Facebook page here. Visit them and add us. Thanks!

NEWS

2012-11-16 - Tutorial on queue call-backs for Elastix using WombatDialer
2012-11-13 - Can Least Cost Routing Exist without VoIP Fraud?
2012-11-12 - Speedflow invites to attend the MediaCore Free Webinar dedicated to the revenue assurance mechanism Guardian
2012-11-12 - Asterisk Channels Live Version 4.0 is available now,more functionality like HangUp,Park Call,Pick Call parked,Call Monitor/Record,Spy & Whisper,Transfer, Transfer to Queue,Transfer to Conference,Originate calls.
2012-11-9 – Apps Connecting Asterisk to Smart Phones PIKA Technologies
2012-11-8 - SIPRoutes Announces Latest Update to its Advanced SIP Termination Least Cost Routing Platform
2012-11-8 - Unicoi Systems Fusion Voice Engine PLC improves Voice Quality for 4G LTE and VoIP!
2012-11-6 - Homer SIP Capture Server with OpenSIPs and Asterisk Spanish HowTo
2012-11-6 -

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Old News

November 17, 2012, 2:24 pm

≫ Next: ICTBroadcast

≪ Previous: voip-info.org

This page lists all the old VoIP news stories from the home page.

Page Contents

October 2012

2012-10-31 - New licensing model for 2N® Helios IP intercoms
2012-10-31 - WombatDialer project update - marching towards an initial release.
2012-10-31 - QueueMetrics 12.10 released - improved scalability and performance - see What's new
2012-10-30 - Humbug Analytics v4.0 Release - New fully updated release of the Humbug Analytics and Fraud Detection platform
2012-10-30 - Edvina launches new SIP Masterclass in Miami, December 2012 - Learn SIP and Kamailio!
2012-10-26 - OrderlyStats SE 10.7 released. Many new features and fully compatible with all known versions of Asterisk, including Elastix and FreePBX.
2012-10-23 - New Hypermedia Distributor in Russia & CIS - Hypermedia Systems has selected Euromobile as its new distributor in Russia and CIS.
2012-10-25 - TransNexus releases BroadSoft compatible SDReporter 4.0 with enhanced fraud detection
2012-10-19 - New VoIPmonitor 5.1 with T.38 to PDF and new issue tracker
2012-10-17 – Mobile Industry Outlook PIKA Technologies
2012-10-16 – Kamailio SIP Server v3.3.2 - new stable release is out
2012-10-09 -

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ICTBroadcast

November 17, 2012, 9:23 pm

≫ Next: ICTDialer

≪ Previous: Old News

ICTBroadcast is web based multi tenant unified communication telemarketing software solution, It features SMS messaging, Email marketing, Fax blasting and Voice broadcasting, suitable for SMB's , Enterprenuers and ITSP. ICTBroadcast supports multiple type of Communication Engines including re-known open source Asterisk, Freeswitch and Kannel. ICT Broadcast is scalbalbe and integerated with RabbitMQ to achieve scalability and it can be scaled to blast thousands of simultaneous voice calls using either VoIP ( SIP or IAX ) or PSTN and Fax calls using using either FOIP (T.38 / G.711 pass through ) or PSTN. . It is simple, reliable and user friendly web portal to manage

ICT Broadcast platform support following type of campaigns

Simple Voice Broadcasting )
Voice Broadcasting with direct forwarding to Live agents support on answer)
Interactive Voice Broadcasting / press 1 campaign )
Survey / Polls )
Inbound IVR campaigns)
SMS Broadcasting )
Fax Broadcasting )
Email Marketing )
Custom IVR voice broadcasting)

How Voice Broadcasting works

User upload a list of telephone numbers, upload audio message or record his voice message through telephone , configure outbound voice gateways and start a new campaign according to requirements using ICT Broadcast web interface and within seconds, ICT Broadcast starts broadcasting user's voice message to given list of telephone numbers with real time statistics.

ICTBroadcast Deployment Scenarios

Automated Telemarketing
Enterprise grade message Broadcasting
Emergency notification system
Interactive voice broadcasting / Smart Predictive dialer
Customer surveys / Collections
Polling Auto Dialler
Mass Communications / notifications
Political voice broadcast
Robocall / call blasting,
Phone reminders
Community / Emergency alerts
School Notifications
Non-profit Fund Raising
Wedding invitations
cold calling,
mass broadcasting.
Appointment reminders
Retail sales / Buisness advertisment

ICT Broadcast Features

For more detail please visit ICTBroadcast web site

ICTBroadcast is developed by ICT Innovations

↧

ICTDialer

November 17, 2012, 9:25 pm

≫ Next: ICTInvoice an open source Elastix module for invoice managment

≪ Previous: ICTBroadcast

ICTDialer , An Open Source Unified Communications auto dialer Software

ICTDialer is open source Unified Communications marketing Software. ICTDialer is multi-tenant with Voice, SMS & Fax broadcasting capabilities developed over re-known open source Content Management System Drupal and Freeswitch based powerful Plivo Communication Framework . It can be scaled to blast thousands of simultaneous calls using either VoIP, Foip or PSTN. ICTDialer capable to fit in many broadcasting and telemarketing scenarios. It empowers user with capabilities of Drupal CMS and Plivo Communication Framework

ICTDialer is released as Open Source GNU GPLv3

ICTDialer is developed and promoted by ICT Innovations with in-depth experience in open source ICT's , Unified Communications and Telemarketing technologies.

↧

ICTInvoice an open source Elastix module for invoice managment

November 17, 2012, 9:33 pm

≫ Next: VOIP Service Providers Business Middle East

≪ Previous: ICTDialer

Introduction

ICTInovice is an Elastix module that enhanses capabilities of Elastix billing and empower elastix admin to automatically generate and email invoices to users of system on monthly basis based on call details records applying rates already set in elastix system and enable users to view or download their invoices in PDF format from elastix web interface. The invoices are based per extension basis which are assigned to users. ICTInvoice has multi tenant capabilities as system admin can manage multiple companies , multiple users per company and multiple extensions per users by assigning / deassigned from web interface, ICTInovice enable admin to manage multiple invoice formats for different companies by creating new templates for each company.

ICTInvoice Ver 2.2.2 released with inbound / outbound billing and user wise summry and detail reporting , for more information please click here ICT Invoice , a multi tenant pbx solution

ICTInvoice is open source GPL v3.0 software, developed and maintained by ICT Innovations and sponsored by Mark Brooker

Installation

Requirment Elastix 1.6 or higher elastix version

Download package file from following link

http://sourceforge.net/projects/ictinvoice

Login your elastix server as admin

Elastix 1.6.x

Click System => Load Module and click on browse button to select package file

Elastix 2.x

You need install “developer module” and then load latest version of ICTInvoice via load module interface

logout / login to activate module

Admin Manual ¶

User Managment

Create users with required privelages and role through elastix menu system => user managment

Extension Mangment

Create required extensions through elastix menu PBX => Extension batch

Ceate New Company Templates

Click Inovoices => Invoice managment => Company templates => create new company template

Assigning users to company Click invoices => Invoice managment => Company Users

Select company name from top

Check un selected users from list

Click on top button to assign / deassing users to company

Assigning extensions to Users

Click Invoices => Invoice mangment => User Extensions

Select user from top

Select un assigned extension from list

Click on top button to assign / deassign extension to users

Create Invoices

Click Invoices => invoice managment => Create inovices

Select company name and user name from top right

Click on button "create invoices" to create invoices

View & Download Invoices.

Click Invoices =? My Invoices

Click on view to view invoice or click on download button to download invoices in PDF format

ICTInvoice is open source GPL v3 software, developed and maintained by ICT Innovations and sponsored by Mark Brooker

Note:

To setup ICTInvoice to offer hosted PBX services to your users, Please visit following link for more detail

↧

VOIP Service Providers Business Middle East

November 20, 2012, 5:52 am

≫ Next: IPsmarx Technology, Inc.

≪ Previous: ICTInvoice an open source Elastix module for invoice managment

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:

VOIP Service Providers Residential Single line residential/business plans go here.
VOIP Service Providers B2B Bulk origination/termination goes here.
RIP VOIP VOIP provider cemetery

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.

Marketing is NOT ALLOWED on this page. 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.

If you like this page, please link to it, so Google and other search engines will consider it more important.

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

↧

IPsmarx Technology, Inc.

November 20, 2012, 9:23 am

≫ Next: Radio

≪ Previous: VOIP Service Providers Business Middle East

Learn more about IPsmarx VoIP Solutions!
We deliver comprehensive and turnkey, customer management and billing solutions for VoIP Service Providers, Carriers, VoIP Calling Card Operators, and ISPs. Our Solutions enable Internet Telephony Service Providers (ITSPs) and entrepreneurs to deploy and run different VoIP business models with all the necessary tools.

Our portfolio includes:

Prepaid and Postpaid Pinless Calling Card Solutions
Softswitch IP-IP Billing Solution
CallShop and Hosted CallShop Solution
Wholesale Carrier Solution
A-Z Wholesale Carrier Services
SIP Enabled Cell Phone Support
Multi Tenant IP PBX
Class 5 softswitch

We take pride in providing highly efficient, modular, scalable, and reliable solutions at very competitive costs that result in rapid return of investments.

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Radio

November 20, 2012, 1:30 pm

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Radio over VoIP

A page to collect resources about linkages between Radios & Voice over IP

Asterisk cmd Rpt: Module for using Asterisk to create a Repeater for Amateur or Commercial 2-way Radios
iaxRPT: A "soft base station" Linux and Windows GUI client derived from iaxComm for the Rpt Asterisk application
DingoTel: Sell commercial '2-way' product for linking 2-way radios to their VoIP service. Mass market approach makes this popular product available online or at Best Buy nationwide.
eQSO: Free software to create Repeaters/Gateways to link 2-way radios across the Internet (either PMR446 or HAM)
Wireless VOIP: Voice over 2.4 GHz and 5Ghz frequencies http://www.shop-wifi.com/index.php?route=product/category&path=33 and http://cordless4u.com/voip.htm
RoA: VoIP AutoPatch
New ROIP solution - Radio with high end options such as Bluetooth, Voip, Skype or GPS : http://cordless4u.com/rexonrl328SBTGPS.htm

General Radio info:

HF

Amateur Radio (licensed)
CB (EU/US licensed) 27 MHz

VHF

Amateur Radio (licensed)

UHF

Amateur Radio (licensed)
PMR446 (EU unlicensed) 446 MHz: http://www.446user.co.uk/ run a network of repeaters linked via eQSO
FRS (US unlicensed) 462-467 MHz: http://f-r-s.org/
GMRS (US licensed) 462-467 MHz: http://g-m-r-s.org/
Australian CBRS (class licensed) 476-477 MHz: more info here

Family Radio Service (FRS) / Commercial Radios / Amateur HAM radios
Professional mobile radios PMR LMR (known as private mobile radio or Land mobile radio systems)
Frequency bands:

L-VHF 26-32MHz, 32-43MHz, 66-88Mhz FM
VHF 135-174Mhz FM , Marine band VHF 156-164Mhz , Air band VHF AM 118-136.975MHz
UHF - 300-360Mhz, 330-400Mhz, 400-470Mhz, 450-527Mhz FM
http://www.shop-wifi.com/index.php?route=product/category&path=17

Thanks to all of you guys for your advices. I already tried to do what was written on the wiki yet I don 't hear any music from the site. Correct me if I was right on the steps I made.

1. I have this on my musiconhold. ...

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SIP Trunk Providers Netherlands

November 21, 2012, 4:25 am

≫ Next: Sip Trunking Providers

≪ Previous: Radio

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

Tritel BV - Affordable, Reliable and Accessible
Voys Telecom - We change business telephony. High quality VoIP Trunks and Hosted VoIP accounts with the best service.

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