Asterisk CTI

Discover the completely open source solution to achieve in a simple and fast system of CTI (Computer Telephony Integration).

The project consists of a client-server solution written in C + + and based on the QT libraries for maximum portability between platforms more widespread.
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Asterisk 1.6 downloader

If you are among those who usually install Asterisk from source, here at your disposal a handy utility that allows you to automatically download the latest stable version of the 1.6 branch.
The utility is dialog-based and downloads the sources in / usr / src / asterisk if executed by root, in ~ / asterisk if performed by non-root user.
Asterisk 1.6 downloader picks, depending on the version selected:

  • Asterisk
  • asterisk-addons
  • libpri
  • dahdi

To use Asterisk 1.6 downloader:

~# wget http://centralino-voip.brunosalzano.com/asterisk16downloader.tar.gz
~# tar zxvf asterisk16downloader.tar.gz
~# chmod u+x asterisk16downloader.sh
~# apt-get install dialog
~# ./asterisk16downloader

Follow the directions on the screen and good downloads!

Files:

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Tags: Call Center Systems | VoIP PBX | Asterisk Consultant Naples | Switchboard Phone | Voip | Asterisk CTI | PBX | IP Phones | Networking | Linux


Development of IVR, call center systems, Voip PBX.

SBNTech VPAD SVP5000

VPAD Video call 1 Video Call 2

Lately I have had the opportunity to try a SIP video phone unique: it is produced by the SVP 5000 SBNTech.

The key features of this product are:

  • Videophone Calls over the Internet
  • LCD Touch Screen (480 × 10.2 pollici/800)
  • Conform to IETF, ITU-T
  • User-friendly interface
  • Video calls @ 30 FPS
  • Digital Photo Frame as a Screen Saver
  • Web Browser
  • SD Card Slot & USB 2.0
  • WiFi (802.11b/g/n)
  • Bluetooth (option)
  • An FXS port
  • Integration HSDPA
  • Video / Audio Input / Output
  • Camera / LCD Power Save Mode
  • Support for audio codecs g711/g729a
  • Support for video codecs H263 and H264 (QCIF / CIF / QVGA / VGA)

The tests were conducted using both Asterisk Asterisk 1.6.x to 1.4.x on CentOS Ubuntu Server in both cases installed from source. The configuration used for the tests was the following:

[general]
videosupport=yes; abilita il supporto video in asterisk
maxcallbitrate=384

[300]
secret=passwordchevuoitu
type=friend
host=dynamic
qualify=yes
context=from-internal
disallow=all
allow=h264
allow=h263
allow=g729
allow=alaw
canreinvite=yes
maxcallbitrate=384

[301]
secret=passwordchevuoitu
type=friend
host=dynamic
qualify=yes
context=from-internal
disallow=all
allow=h264
allow=h263
allow=g729
allow=alaw
canreinvite=yes
maxcallbitrate=384

In principle, I suggest you avoid, as far as possible a setup where you have to worry about Asterisk retransmit the audio / video streams to the two end-points. This is the reason why, in the suggested configuration, the parameter "canreinvite" is set to "yes".

The parameter "canreinvite = yes" makes it possible to Asterisk, once the call has been accepted, to send a new INVITE to the two endpoints with the information necessary for establishment of a audio / video communication directly between them.

Please note that this "reinvite" does not happen, despite the "canreinvite = yes" if:

1) one of the two clients is configured with "canreinvite = no";

2) if the clients are using different codecs.

3) If the command Dial () contains the parameters for recording sound (wo W) / call transfer (to T) / forced hangup (I H).

In all cases, the bandwidth required for a video call and of at least 200Kbps, always that the canreinvite is active. In other cases need more bandwidth. Attention to the fact that as soon as an active video call, the phones use a value of bandwidth as low as possible. To get higher quality video you must manually increment the bandwidth parameter dall'apposito configuration panel in the call.

In my testing I used: H264 / G729 / VGA. It 's always possible to scale the parameter VGA to QVGA or QCIF if you want to further decrease (although not by much) the required bandwidth.

The tests carried out also seems that only when the work canreinvite a video call between the phone and a desk phone is not activated in the first video. In other cases, it goes hand-chosen not to activate the video when you make or accept a video call from the phone.

Very intuitive and simple user interface: All functions are easily accessible thanks to the convenient touch screen.

The product was kindly provided by www.4geek.it

Manufacturer's page: SBNTech

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Development of IVR, call center systems, Voip PBX.

Porting GTK Asterisk CTI

Italian Version "
astcti_gtk.jpg astcti_gtk_lin.jpg

I started a few weeks from a general restructuring of Asterisk CTI. The new version, For which is indicatively planned for release in early 2009 Represents a revolution against the first version of the project.

The main features are:

  • Approach completely renovated: the Centralized server configuration and client services Entitled to be Notified of calls, and applications to be Launched on workstations.
  • Software completely rewritten in C # with mono, using the GTK libraries. This will have a truly cross-platform software.
  • Client, called AstCTIClient, rewritten GTK and multiplatform.
  • All configuration can be handled by software called AstCTIConfigurator and will be stored in a MySQL database.
  • The configurations will be saved in an XML file to be loaded at AstCTIServer: This will make the server independent of any failures of the MySQL database.
  • Multilanguage support.
Of course this is an Open Source project under GPL and short will upload the sources on a special project page on google code. I await your comments and feedback, especially a stake in the project!
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Development of IVR, call center systems, Voip PBX.

Porting GTK Asterisk CTI

Inglese Version
astcti_gtk.jpg astcti_gtk_lin.jpg

I started a few weeks ago a general restructuring of Asterisk CTI. The new version, for which a release is tentatively scheduled for early 2009, is a revolution compared to the first version of the project.

The main changes are:

  • Approach completely renewed centralization on the server of the service configuration and enabled clients to be notified of calls, as well as the applications to run on workstations.
  • Software completely rewritten in C # with mono using GTK. This will be a truly multi-platform software.
  • Client called AstCTIClient, rewritten GTK and multiplatform.
  • All configuration can be managed by software called AstCTIConfigurator. and will be stored in a MySQL database.
  • The configurations will be saved in an XML file loaded at startup from AstCTIServer: this will make the server independent of any failures of the MySQL database.
  • Multilanguage support.

Of course it is an open source project licensed under the GPL and I will soon upload the source code on a special project page on google code. I await your comments, feedback and, above all, a participation in the project!

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Development of IVR, call center systems, Voip PBX.

Asterisk Blog

Here I am again to update the pages of this blog asterisk after a period of inactivity, mainly due to the workload of the previous months. During this period I had the opportunity to not only enhance experiences in the field of telephony, but also to make new and important knowledge.

Some major changes await the pages of this blog, I list a few:

  • Constant updates of news
  • New technical articles on configuring Asterisk
  • Articles on programming AGI and AMI
  • Review of products related to the world of telephony

I expect many of the pages of my blog!

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Development of IVR, call center systems, Voip PBX.

Upcoming release of AsteriskCTI

Coming soon some improvements in the software AsteriskCTI :

  • Localization support with possibility of adding simply new translations
  • Ability to disable the client on the user configuration
  • Ability to disable the client on the outbound
  • Integrated Web browser for better management of web-applications CTI. The browser has a basic level of configuration, it is possible to implement several functions on request.
  • Support for agents on the server side
  • Support at work after call (PostCall Work) for the campaign.

At the moment some of these features have already been released in ' SVN but are still in beta.

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Development of IVR, call center systems, Voip PBX.

Asterisk 1.4.21 Released

The Asterisk development team has released version 1.4.21.

This is a normal release aims to fix bugs. For a complete list of changes please refer to the file ChangeLog included in the release.

Asterisk 1.4.21 is available for download from Digium:

http://downloads.digium.com/pub/telephony/asterisk/

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Development of IVR, call center systems, Voip PBX.

From Asternic Call Center Statistics in real time for Asterisk

Nicolas Gudino - the author of the FOP, the famous Flash Operator Panel - has released a new software for Asterisk designed to monitor the queue statistics called Asternic Callcenter Stats

An excerpt from the home page translated into Italian:

Asternic Call Center Stats is a package to analyze the files queue_log of your Asterisk PBX via a web page and display information in realtim.

There are several reports with beautiful graphics in flash and the option to export to pdf and csv for Excel.

Are required:

On the server

  • Web server with PHP support
  • MySQL Database

On the client

  • A browser with Javascript support enabled
  • Flash Plugin

There are two versions: a free licensed with the GPL v3 version and a commercial version includes many comerciale.La pù reports, different levels of user access, real-time parsing of the file queue_log, the ability to listen to recordings of calls through the streaming.

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Development of IVR, call center systems, Voip PBX.

Loquendo TTS integration

Loquendo TTS software is a Text To Speech mall that includes items for many languages ​​such as English, French, German, Spanish, Italian, Portuguese, Greek, Catalan and Swedish.

The user RPR published on voip-info.org an interesting wiki page where they are shown the various techniques of integration of Loquendo with Asterisk.

The methods below are two basic and alternative are:

  1. A script called loquendo.agi AGI to be included in the directory /var/lib/asterisk/agi-bin . This script is integrated with an executable written in C code and link text2audio.c called to fill in the directory /opt/Loquendo/LTTS7/samples/c/txt2audio .
  2. A dialplan application called app_loquendo. This solution, however, does not seem to work with Asterisk 1.4 or Loquendo 7.
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Development of IVR, call center systems, Voip PBX.

Asterisk 1.4.20-rc2

The development team has released version of Asterisk 1.4.20-rc2.

This version is a "release candidate" for the next stable release 1.4.20. Among other changes, this release includes a number of improvements in the SIP channel driver than the version introduced with the 1.4.20-rc1.

For a complete list of changes please see the ChangeLog file that is distributed with this release.

Links:

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Development of IVR, call center systems, Voip PBX.

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