Asterisk CTI

Check out the completely open source solution to realize 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 library for maximum portability across popular platforms.
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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 based on dialog and download the sources in / usr / src / asterisk if run as root, in ~ / asterisk if performed by non-root user.
Asterisk 1.6 downloader picks, depending on the version:

  • 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 download!

Files:

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


Development of IVR systems, call center, PBX VoIP.

SBNTech VPAD SVP5000

VPAD Video call 1 Video call 2

Lately I've got to try a SIP videophone unique: it is produced by the SVP 5000 SBNTech.

The key features of this product are:

  • Videophone Calls over the Internet
  • Touch Screen LCD (10.2 inches / 800 × 480)
  • Compliance industry standards IETF, ITU-T
  • User-Friendly
  • 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)
  • A FXS port
  • Integration HSDPA
  • Video / Audio Input / Output
  • Room / LCD Power Save Mode
  • Support for audio codecs g711 / G729a
  • Support for video codecs h263 and h264 (QCIF / CIF / QVGA / VGA)

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

[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 suggest you avoid, as far as possible a configuration where Asterisk should worry about retransmit the audio / video streams at the two end-points. This is 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 answered, to send a new INVITE message to the two endpoints with the necessary information to the establishment of a audio / video communication directly between them.

Beware that this "reinvite" will not happen, despite "canreinvite = yes" if:

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

2) if the clients use different codecs.

3) If the command Dial () contains 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, provided 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 must manually increment the parameter bandwidth dall'apposito configuration panel in call.

In my tests 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 call between the video phone and a desk phone is not active in the first video. In other cases, it should be chosen by hand not to activate the video when you make or accept a call from the video phone.

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

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

Manufacturer's page: SBNTech

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Development of IVR systems, call center, PBX VoIP.

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 in 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 the 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 systems, call center, PBX VoIP.

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, represents a revolution compared to the first version of the project.

The main new features are:

  • Approach completely renovated: centralization on the server configuration services 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 the GTK libraries. This will have a truly multi-platform software.
  • Client, called AstCTIClient, rewritten in GTK and multiplatform.
  • All configuration can be managed by a 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 under the GPL and short I will upload the source code on a special page of the project on google code. I await your comments, feedback and, above all, a participation in the project!

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Development of IVR systems, call center, PBX VoIP.

Asterisk Blog

I'm back to update the pages of this blog asterisk after a period of inactivity, mainly due to the workload of the previous months. During this time I was able not only to increase experience in telephony, but also to make new and important knowledge.

Some important 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 systems, call center, PBX VoIP.

Next release AsteriskCTI

Coming soon some improvements in the software AsteriskCTI :

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

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

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Development of IVR systems, call center, PBX VoIP.

Asterisk 1.4.21 released

The Asterisk development team has released version 1.4.21.

This is a normal release aimed to fix bugs. For a complete list of changes, see 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 systems, call center, PBX VoIP.

From Asternic Statistics Call Center in real time for Asterisk

Nicolas Gudino - the author of 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 in Italian:

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

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

Are required:

On the server

  • Webserver with PHP support
  • MySQL Database

On clients

  • A browser with Javascript support enabled
  • Flash plugin

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

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Development of IVR systems, call center, PBX VoIP.

Integration Loquendo TTS

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

You RPR published on voip-info.org an interesting wiki page where describes the various techniques of integration of Loquendo with Asterisk.

The methods below are basically two and are alternatives:

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

Asterisk 1.4.20-rc2

The development team has released version 1.4.20-rc2 Asterisk.

This version is a "release candidate" for the next release of the stable version 1.4.20. Among other changes, this version 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, see the ChangeLog file that is distributed with this release.

Useful links:

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Development of IVR systems, call center, PBX VoIP.

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