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Math::Telephony::ErlangB 0.06
Math::Telephony::ErlangB is a Perl extension for Erlang B calculations. more>>
Math::Telephony::ErlangB is a Perl extension for Erlang B calculations.
SYNOPSIS
use Math::Telephony::ErlangB qw( :all );
# Evaluate blocking probability
$bprob = blocking_probability($traffic, $servers);
$gos = gos($traffic, $servers); # Same result as above
# Dimension minimum number of needed servers
$servers = servers($traffic, $gos);
# Calculate maximum serveable traffic
$traffic = traffic($servers, $gos); # Default precision 0.001
$traffic = traffic($servers, $gos, 1e-10);
This module contains various functions to deal with Erlang B calculations.
The Erlang B model allows dimensioning the number of servers in a M/M/S/0/inf model (Kendall notation):
- The input process is Markovian (Poisson in this case)
- The serving process is Markovian (ditto)
- There are S servers
- Theres no wait line (pure loss)
- The input population is infinite
<<lessSYNOPSIS
use Math::Telephony::ErlangB qw( :all );
# Evaluate blocking probability
$bprob = blocking_probability($traffic, $servers);
$gos = gos($traffic, $servers); # Same result as above
# Dimension minimum number of needed servers
$servers = servers($traffic, $gos);
# Calculate maximum serveable traffic
$traffic = traffic($servers, $gos); # Default precision 0.001
$traffic = traffic($servers, $gos, 1e-10);
This module contains various functions to deal with Erlang B calculations.
The Erlang B model allows dimensioning the number of servers in a M/M/S/0/inf model (Kendall notation):
- The input process is Markovian (Poisson in this case)
- The serving process is Markovian (ditto)
- There are S servers
- Theres no wait line (pure loss)
- The input population is infinite
Download (0.024MB)
Added: 2006-07-04 License: Perl Artistic License Price:
1210 downloads
Math::Telephony::ErlangC 0.03
Math::Telephony::ErlangC is a Perl extension for Erlang C calculations. more>>
Math::Telephony::ErlangC is a Perl extension for Erlang C calculations.
SYNOPSIS
use Math::Telephony::ErlangC;
# Evaluate probability that a service request will have to wait
$wprob = wait_probability($traffic, $servers);
# Probability that the wait time will be less than a fixed maximum
# $mst is the mean service time
$mwprob = maxtime_probability($traffic, $servers, $mst, $maxtime);
# Average time waiting in queue
# $mst is the mean service time
$awtime = average_wait_time($traffic, $servers, $mst);
This mosule contains various functions to deal with Erlang C calculations.
The Erlang C model allows dimensioning the number of servers in a M/M/S/inf/inf model (Kendall notation):
- The input process is Markovian (Poisson in this case)
- The serving process is Markovian (ditto)
- There are S servers
- The wait line is infinite (pure wait, no loss, no renounce)
- The input population is infinite
<<lessSYNOPSIS
use Math::Telephony::ErlangC;
# Evaluate probability that a service request will have to wait
$wprob = wait_probability($traffic, $servers);
# Probability that the wait time will be less than a fixed maximum
# $mst is the mean service time
$mwprob = maxtime_probability($traffic, $servers, $mst, $maxtime);
# Average time waiting in queue
# $mst is the mean service time
$awtime = average_wait_time($traffic, $servers, $mst);
This mosule contains various functions to deal with Erlang C calculations.
The Erlang C model allows dimensioning the number of servers in a M/M/S/inf/inf model (Kendall notation):
- The input process is Markovian (Poisson in this case)
- The serving process is Markovian (ditto)
- There are S servers
- The wait line is infinite (pure wait, no loss, no renounce)
- The input population is infinite
Download (0.008MB)
Added: 2006-12-01 License: Perl Artistic License Price:
1057 downloads
Telephony::CountryDialingCodes 1.02
Telephony::CountryDialingCodes is a Perl module that can convert international dialing codes to country codes and vice versa. more>>
Telephony::CountryDialingCodes is a Perl module that can convert international dialing codes to country codes and vice versa.
SYNOPSIS
# Usage method 1 (using object methods):
use Telephony::CountryDialingCodes;
my $o = new Telephony::CountryDialingCodes();
my $country_code = NL;
print "The dialing access code for country $country_code is " . $o->dialing_code($country_code) . "n";
my $dialing_code = 1;
my @country_codes = $o->country_codes($dialing_code);
print "The country code(s) for dialing access code $dialing_code is/are: " . join(,,@country_codes) . "n";
# Usage method 2 (using class methods):
use Telephony::CountryDialingCodes;
my $country_code = NL;
print "The dialing access code for country $country_code is " . Telephony::CountryDialingCodes->dialing_code($country_code) . "n";
my $dialing_code = 1;
my @country_codes = Telephony::CountryDialingCodes->country_codes($dialing_code);
print "The country code(s) for dialing access code $dialing_code is/are: " . join(,,@country_codes) . "n";
# Extracting an intl dialing code from an intl phone number:
use Telephony::CountryDialingCodes;
my $o = new Telephony::CountryDialingCodes();
my $dialing_code = $o->extract_dialing_code(+521234567890);
# $dialing_code will contain 52.
This class exports a method for determining a countrys international dialing code, and another method for doing the reverse: i.e. determining the country code(s) that belong(s) to a given international dialing code.
You can call these methods as class methods or you can create an object and call these methods as object methods. The difference is that if you call them in object context that the internal lookup tables are freed when the object is destroyed, otherwise if you call the methods in class context, then the internal lookup tables are global and will persist for the lifespan of the current process. Its not really a big deal which approach you choose, so for the sake of style, use the object method approach if you have no clue which is better.
<<lessSYNOPSIS
# Usage method 1 (using object methods):
use Telephony::CountryDialingCodes;
my $o = new Telephony::CountryDialingCodes();
my $country_code = NL;
print "The dialing access code for country $country_code is " . $o->dialing_code($country_code) . "n";
my $dialing_code = 1;
my @country_codes = $o->country_codes($dialing_code);
print "The country code(s) for dialing access code $dialing_code is/are: " . join(,,@country_codes) . "n";
# Usage method 2 (using class methods):
use Telephony::CountryDialingCodes;
my $country_code = NL;
print "The dialing access code for country $country_code is " . Telephony::CountryDialingCodes->dialing_code($country_code) . "n";
my $dialing_code = 1;
my @country_codes = Telephony::CountryDialingCodes->country_codes($dialing_code);
print "The country code(s) for dialing access code $dialing_code is/are: " . join(,,@country_codes) . "n";
# Extracting an intl dialing code from an intl phone number:
use Telephony::CountryDialingCodes;
my $o = new Telephony::CountryDialingCodes();
my $dialing_code = $o->extract_dialing_code(+521234567890);
# $dialing_code will contain 52.
This class exports a method for determining a countrys international dialing code, and another method for doing the reverse: i.e. determining the country code(s) that belong(s) to a given international dialing code.
You can call these methods as class methods or you can create an object and call these methods as object methods. The difference is that if you call them in object context that the internal lookup tables are freed when the object is destroyed, otherwise if you call the methods in class context, then the internal lookup tables are global and will persist for the lifespan of the current process. Its not really a big deal which approach you choose, so for the sake of style, use the object method approach if you have no clue which is better.
Download (0.004MB)
Added: 2006-09-20 License: Perl Artistic License Price:
1130 downloads
Yet Another Telephony Engine 1.2.0
YATE is a telephony engine aimed at creating an telephony server. more>>
Voice, video, data and instant messaging can all be unified under Yates flexible routing engine, maximizing communications efficiency and minimizing infrastructure costs for businesses.
Main features:
- VoIP server
- VoIP client
- VoIP to PSTN gateway
- PC2Phone and Phone2PC gateway
- H.323 gatekeeper
- H.323 multiple endpoint server
- SIP session border controller
- SIP router
- SIP registration server
- IAX server and client
- IP Telephony server and client
- Call center server
- IVR engine
- Prepaid and postpaid cards system
The software is written in C++ and it supports scripting in various programming languages (such as those supported by the curently implemented embedded PHP and Python interpreters).
Both PHP and Python libraries have been developed and made available in order to ease development of external functionalities for Yate.
Yate is production-ready software.
Yates license is GPL with an exception for linking with OpenH323 and PWlib (licensed under MPL).
Enhancements:
- This release adds support for Jingle (GoogleTalk), XML, and improved PBX support.
- It fixes one minor security bug and several others not security related.
<<lessMain features:
- VoIP server
- VoIP client
- VoIP to PSTN gateway
- PC2Phone and Phone2PC gateway
- H.323 gatekeeper
- H.323 multiple endpoint server
- SIP session border controller
- SIP router
- SIP registration server
- IAX server and client
- IP Telephony server and client
- Call center server
- IVR engine
- Prepaid and postpaid cards system
The software is written in C++ and it supports scripting in various programming languages (such as those supported by the curently implemented embedded PHP and Python interpreters).
Both PHP and Python libraries have been developed and made available in order to ease development of external functionalities for Yate.
Yate is production-ready software.
Yates license is GPL with an exception for linking with OpenH323 and PWlib (licensed under MPL).
Enhancements:
- This release adds support for Jingle (GoogleTalk), XML, and improved PBX support.
- It fixes one minor security bug and several others not security related.
Download (0.83MB)
Added: 2007-04-16 License: GPL (GNU General Public License) Price:
922 downloads
SFLphone 0.7
SFLphone is a Voice over IP telephony software. more>>
SFLphone is a Voice over IP telephony software.
Voice over IP tends to be part of more and more corporate (and home) networks today.
As Open Source Software become more and more popular in companies, we decided to provide the community with a new powerful and open platform for desktop-based Voice over IP. SFLphone aims to become your desktops VoIP companion.
Main features:
- SIP and STUN support ;
- OSS and ALSA (since 0.3) audio devices support ;
- Skinned GUI ;
- G711u, G711a PCM, and GSM (since 0.3) codecs ;
- And nice other features !
<<lessVoice over IP tends to be part of more and more corporate (and home) networks today.
As Open Source Software become more and more popular in companies, we decided to provide the community with a new powerful and open platform for desktop-based Voice over IP. SFLphone aims to become your desktops VoIP companion.
Main features:
- SIP and STUN support ;
- OSS and ALSA (since 0.3) audio devices support ;
- Skinned GUI ;
- G711u, G711a PCM, and GSM (since 0.3) codecs ;
- And nice other features !
Download (0.47MB)
Added: 2006-09-13 License: GPL (GNU General Public License) Price:
1139 downloads
YateClient 1.2.0
YateClient is a next-generation telephony engine. more>>
YateClient is a next-generation telephony engine.
While currently focused on Voice over Internet Protocol (VoIP) and PSTN, its power lies in its ability to be easily extended.
Voice, video, data, and instant messaging can all be unified under Yates flexible routing engine, maximizing communications efficiency and minimizing infrastructure costs for businesses.
Yate can be used as a:
- VoIP server
- VoIP client
- VoIP to PSTN gateway
- PC2Phone and Phone2PC gateway
- H.323 gatekeeper
- H.323 multiple endpoint server
- H.323SIP Proxy
- SIP session border controller
- SIP router
- SIP registration server
- IAX server and/or client
- IP Telephony server and/or client
- Call center server
- IVR engine
- Prepaid and/or postpaid cards system
<<lessWhile currently focused on Voice over Internet Protocol (VoIP) and PSTN, its power lies in its ability to be easily extended.
Voice, video, data, and instant messaging can all be unified under Yates flexible routing engine, maximizing communications efficiency and minimizing infrastructure costs for businesses.
Yate can be used as a:
- VoIP server
- VoIP client
- VoIP to PSTN gateway
- PC2Phone and Phone2PC gateway
- H.323 gatekeeper
- H.323 multiple endpoint server
- H.323SIP Proxy
- SIP session border controller
- SIP router
- SIP registration server
- IAX server and/or client
- IP Telephony server and/or client
- Call center server
- IVR engine
- Prepaid and/or postpaid cards system
Download (1.1MB)
Added: 2007-04-16 License: GPL (GNU General Public License) Price:
923 downloads
WengoPhone 2.1.1
WengoPhone is a multi-platform VOIP client sponsored and developed by WENGO and MBDSYS. more>>
The Wengophone is an easy to use program, that enables you to call anyone, anywhere, anytime, for free. Its native use of the SIP protocol makes it interoperable with most on the known VoIP technologies.
WengoPhone already has built-in video capabilities, and is will offer many more features in the near future such as calling ordinary phones (land lines and mobiles) at very low prices, conference calls, user communities and innovative cellphone-based services.
With our software, you have better audio quality, broader connectivity and low bandwidth usage. Additionally, thanks to its GPL licence and SIP-compliant foundations, you get the freedom of open source software: add functionnality or modify Wengophone to fit your needs.
We are aware of the challenge that is launching an open source project, not just in terms of creating a new technology, but also in getting developers and future customers to support the efforts. We are commited to consistent and honest communication.
OpenWengo and Wengos commercial ventures are complementary, and our goal is to provide an alternative to expensive and/or proprietary services.
We expect to compete strongly with Skype in the realm of software-based services (PC to PC) and complementary access modes (ATA, mobile devices). The trick? Free calls, Free software.
SIP, short for Session Initiation Protocol, is an open standard developed by the Internet Engineering Task Force. Eventually, it will enable interoperability with other compliant VoIP service providers. One will be able to call any other SIP-compliant softphone user, seamlessly and for free. Moreover, SIP will enable new services for mobile users.
Main features:
- MakingCalls
- ChatSupport
- VideoSupport
- Presence
- EmotiSounds
- FileTransfers
Wengo uses your broadband internet connection (DSL, Cable or WiFi) to provide telephony services. Call your friends, family, and colleagues anywhere, anytime, for free. Use video and chat features, also for free.
Wengo also offers low cost international rates (prepaid or unlimited) to land lines or mobile phones, and an SMS gateway service.
Wengo provides its customers with a Wenbox ATA, that enables the use of any regular phone. This device converts analog phone signals so that calls can be placed over the internet.
<<lessWengoPhone already has built-in video capabilities, and is will offer many more features in the near future such as calling ordinary phones (land lines and mobiles) at very low prices, conference calls, user communities and innovative cellphone-based services.
With our software, you have better audio quality, broader connectivity and low bandwidth usage. Additionally, thanks to its GPL licence and SIP-compliant foundations, you get the freedom of open source software: add functionnality or modify Wengophone to fit your needs.
We are aware of the challenge that is launching an open source project, not just in terms of creating a new technology, but also in getting developers and future customers to support the efforts. We are commited to consistent and honest communication.
OpenWengo and Wengos commercial ventures are complementary, and our goal is to provide an alternative to expensive and/or proprietary services.
We expect to compete strongly with Skype in the realm of software-based services (PC to PC) and complementary access modes (ATA, mobile devices). The trick? Free calls, Free software.
SIP, short for Session Initiation Protocol, is an open standard developed by the Internet Engineering Task Force. Eventually, it will enable interoperability with other compliant VoIP service providers. One will be able to call any other SIP-compliant softphone user, seamlessly and for free. Moreover, SIP will enable new services for mobile users.
Main features:
- MakingCalls
- ChatSupport
- VideoSupport
- Presence
- EmotiSounds
- FileTransfers
Wengo uses your broadband internet connection (DSL, Cable or WiFi) to provide telephony services. Call your friends, family, and colleagues anywhere, anytime, for free. Use video and chat features, also for free.
Wengo also offers low cost international rates (prepaid or unlimited) to land lines or mobile phones, and an SMS gateway service.
Wengo provides its customers with a Wenbox ATA, that enables the use of any regular phone. This device converts analog phone signals so that calls can be placed over the internet.
Download (MB)
Added: 2007-06-28 License: GPL (GNU General Public License) Price:
1611 downloads
Qtel 0.90
Qtel stands for the QT EchoLink client. more>>
Qtel stands for the "QT EchoLink" client. It is only a EchoLink client application. There is no "sysop mode". If it is a link you want to run, have a look at the SvxLink (http://svxlink.sourceforge.net) server application.
EchoLink is an amateur radio invention (well actually it is just a modified verison of IP telephony) to link radio transcievers together over the Internet. You must have an amateur radio license to use it.
The original EchoLink software can be found at http://www.echolink.org/. However, this software only supports the Windows operating system and it is closed source. SvxLink is released under the GPL license. Qtel is only a EchoLink client program.
<<lessEchoLink is an amateur radio invention (well actually it is just a modified verison of IP telephony) to link radio transcievers together over the Internet. You must have an amateur radio license to use it.
The original EchoLink software can be found at http://www.echolink.org/. However, this software only supports the Windows operating system and it is closed source. SvxLink is released under the GPL license. Qtel is only a EchoLink client program.
Download (0.10MB)
Added: 2005-08-31 License: GPL (GNU General Public License) Price:
1529 downloads
YateAdmin 1
YateAdmin is a FREE powerful web interface which helps to set up the telephony engine Yate. more>>
YateAdmin is a FREE powerful web interface which helps to set up the telephony engine Yate.
You can use it to setup:
- Users lines
- Register to other servers
- Routes for gateways
- Fallback routing
- Call Detail Report
- Yate status
<<lessYou can use it to setup:
- Users lines
- Register to other servers
- Routes for gateways
- Fallback routing
- Call Detail Report
- Yate status
Download (0.058MB)
Added: 2006-10-04 License: GPL (GNU General Public License) Price:
1120 downloads
PPhone 1.5.0 Beta 4
PPhone is a universal communication client for IP telephony and instant messaging. more>>
PPhone project is a universal communication client for IP telephony and instant messaging.
The applications design is modular, so GUI, protocol, codec, and sound drivers have plugin interfaces, the GUI is detachable, and you can switch among different GUI implementations and change the clients configuration at runtime.
With the gaimhack module, you can use IM plugins from Gaim 2 to access mainstream IM networks.
Enhancements:
- The STUN interface and module have been released.
- SIP WWW-Authorization support has been added.
- There are curses contact list control fixes and minor layout improvements.
<<lessThe applications design is modular, so GUI, protocol, codec, and sound drivers have plugin interfaces, the GUI is detachable, and you can switch among different GUI implementations and change the clients configuration at runtime.
With the gaimhack module, you can use IM plugins from Gaim 2 to access mainstream IM networks.
Enhancements:
- The STUN interface and module have been released.
- SIP WWW-Authorization support has been added.
- There are curses contact list control fixes and minor layout improvements.
Download (2.6MB)
Added: 2007-04-12 License: BSD License Price:
925 downloads
Asterisk Desktop Manager 0.9
Asterisk Desktop Manager aims to integrate your desktop with the Asterisk PBX and hardware IP phone. more>>
Asterisk Desktop Manager application interfaces your VOIP telephony with the desktop and Asterisk PBX.
Asterisk Desktop Manager aims to integrate your desktop with the Asterisk PBX and hardware IP phone by providing some useful features such as automatic on-call volume reduction, one click dialling (from the clipboard), BlueTooth presence detection and automatic call redirection when you walk out of the office, a pop up browser on call, and transfer of the current call from the desktop.
Enhancements:
- Automatic on-call volume reduction
- One click dial from clipboard (paste number onto tray icon)
- Integrated phonebook
- List/Redirect/Hangup all active calls
- One click call forward setup
- Bluetooth presence detection to redirect calls when you walk out of the office
- Pop up browser on incoming call (integrate with your CRM to auto load customers details when they call)
- Cisco phone integration (auto speakerphone)
- Slide-in popup on incoming call, with Answer(cisco only), Hold, Busy and Redirect buttons , CallerID and duration
<<lessAsterisk Desktop Manager aims to integrate your desktop with the Asterisk PBX and hardware IP phone by providing some useful features such as automatic on-call volume reduction, one click dialling (from the clipboard), BlueTooth presence detection and automatic call redirection when you walk out of the office, a pop up browser on call, and transfer of the current call from the desktop.
Enhancements:
- Automatic on-call volume reduction
- One click dial from clipboard (paste number onto tray icon)
- Integrated phonebook
- List/Redirect/Hangup all active calls
- One click call forward setup
- Bluetooth presence detection to redirect calls when you walk out of the office
- Pop up browser on incoming call (integrate with your CRM to auto load customers details when they call)
- Cisco phone integration (auto speakerphone)
- Slide-in popup on incoming call, with Answer(cisco only), Hold, Busy and Redirect buttons , CallerID and duration
Download (9.1MB)
Added: 2006-01-11 License: GPL (GNU General Public License) Price:
787 downloads
Asterisk Live! CD 0.1.6
Asterisk PBX is Linux based, open source PBX software that provides voice over IP in three protocols. more>>
Asterisk PBX is Linux based, open source PBX software that provides voice over IP in three protocols and is interoperable with most standards-based telephony equipment using comparatively inexpensive hardware.
If you want to play around with Asterisk check out this distribution which is available as a Live CD and a Compact Flash install.
<<lessIf you want to play around with Asterisk check out this distribution which is available as a Live CD and a Compact Flash install.
Download (36.4MB)
Added: 2005-05-18 License: GPL (GNU General Public License) Price:
910 downloads
c2cWebdialer 1.0
c2cWebdialer provides an extension which works with a CISCO IP telephony solution to provide an easy way to dial phone numbers. more>>
c2cWebdialer provides an extension which works with a CISCO IP telephony solution to provide an easy way to dial phone numbers.
C2cWebdialer extension works with a CISCO IP telephony solution to provide an easy way to dial phone numbers on web pages. Just select the phone number and choose "Dial with CISCO Webdialer" from the contextual menu !
C2CWebdialer has to be configured through the "Click2Call parameters" menu, it needs two parameters :
* The IP address or the DNS name of the CISCO CallManager on which the webdialer service is launched
* The dial out prefix that C2cWebdialer will put before the selected phone number
The webdialer feature must be activated on the CISCO CallManager.
<<lessC2cWebdialer extension works with a CISCO IP telephony solution to provide an easy way to dial phone numbers on web pages. Just select the phone number and choose "Dial with CISCO Webdialer" from the contextual menu !
C2CWebdialer has to be configured through the "Click2Call parameters" menu, it needs two parameters :
* The IP address or the DNS name of the CISCO CallManager on which the webdialer service is launched
* The dial out prefix that C2cWebdialer will put before the selected phone number
The webdialer feature must be activated on the CISCO CallManager.
Download (0.004MB)
Added: 2007-04-12 License: MPL (Mozilla Public License) Price:
927 downloads
Asterisell 0.9.3
Asterisell project is a Web based application for viewing, rating, and billing Asterisk VoIP calls. more>>
Asterisell project is a Web based application for viewing, rating, and billing Asterisk VoIP calls.
Usage Scenario:
- you are a vendor of Voice over IP Telephony services (VoIP)
- your customers call other users who can reside on different telephone networks
- your customers pay you for this service
in order to route the calls on particular networks you have to use the services of other vendors
- you are a customer of others vendors
- you pays others vendors for this service
So Asterisell views you (the VoIP service provider) as a reseller of others vendors services. It permits to associate for every call a cost and an income. The cost is what the customer pays to the service provider and the cost is what the service provider pays to other vendors in order to route the call.
Main features:
Customer features:
- report of calls with aggregate totals
- many filter-criteria
- different sort-criteria
- export to CSV and MS Excel document
- many VoIP accounts associated with the same customer
Administrator features:
- input fields are secure from intrusion attempts
- each customer could have a cost limit for the last 30 days
- informal currency conversion table for the test of cost limits
- administrator is advised via email if a customer pass its cost limit
- different cultures for each VoIP account or Customer
- prices and messages are displayed in the specified culture
- customers can be grouped into different price list categories
- different rate methods applicable to different time periods, customer categories, destination gateway channels, destination telephone network etc..
- customizable rate methods adding PHP Class respecting a specific interface
- generation of customer invoices
- delivery of emails with invoice details to customers
- support bulk generation of invoices and emails to all customers with associated incomes
- support for multi-currency invoices (not yet tested)
- problems during rate process are reported and signaled via email to the administrator
General features:
- multi-culture and locales support (it_IT, en_US, ecc..)
- multi-currency support (not yet tested)
<<lessUsage Scenario:
- you are a vendor of Voice over IP Telephony services (VoIP)
- your customers call other users who can reside on different telephone networks
- your customers pay you for this service
in order to route the calls on particular networks you have to use the services of other vendors
- you are a customer of others vendors
- you pays others vendors for this service
So Asterisell views you (the VoIP service provider) as a reseller of others vendors services. It permits to associate for every call a cost and an income. The cost is what the customer pays to the service provider and the cost is what the service provider pays to other vendors in order to route the call.
Main features:
Customer features:
- report of calls with aggregate totals
- many filter-criteria
- different sort-criteria
- export to CSV and MS Excel document
- many VoIP accounts associated with the same customer
Administrator features:
- input fields are secure from intrusion attempts
- each customer could have a cost limit for the last 30 days
- informal currency conversion table for the test of cost limits
- administrator is advised via email if a customer pass its cost limit
- different cultures for each VoIP account or Customer
- prices and messages are displayed in the specified culture
- customers can be grouped into different price list categories
- different rate methods applicable to different time periods, customer categories, destination gateway channels, destination telephone network etc..
- customizable rate methods adding PHP Class respecting a specific interface
- generation of customer invoices
- delivery of emails with invoice details to customers
- support bulk generation of invoices and emails to all customers with associated incomes
- support for multi-currency invoices (not yet tested)
- problems during rate process are reported and signaled via email to the administrator
General features:
- multi-culture and locales support (it_IT, en_US, ecc..)
- multi-currency support (not yet tested)
Download (2.5MB)
Added: 2007-06-28 License: GPL (GNU General Public License) Price:
848 downloads

PBXware SOHO 1.8.0
PBXware is a scalable PBX solution. more>> PBXware is a scalable PBX solution featuring a range of traditional telephony and emerging VOIP technologies. The creation of national/global voice networks and a full range of PSTN and VOIP technologies is supported. Least cost routing, Enhanced Voicemail, ACD Queues, IVR Auto Attendants, Conference Bridges , Music on Hold and much more... a real cost saving in a fully featured PBX solution!
PBXware is delivered in CD, tarball, appliance or VPS formats each supporting an easy to use Setup Wizard allowing setup of a fully functional PBX in minutes. Administration is performed through included Web Interface or CLI. Auto updates, system backup, provider templates, call recordings, real time call monitoring are just some of advanced features included.
PBXware offers flexibility, performance, delivery methods, technologies, easy to use yet extremely advanced features not found at any competitors offerings. Should you found same features set elsewhere we would like to hear from you.<<less
Download (700.00MB)
Added: 2009-04-09 License: Freeware Price: Free
205 downloads
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