Voip Pbx

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If you are among those who will 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 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 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 instructions on the screen and good downloads!

Files:

• Asterisk Downloader 1.6

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

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

The key features of this product are:

The tests were conducted using both Asterisk 1.4.x on CentOS that Asterisk 1.6.x on Ubuntu Server in both cases installed by springs. 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 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 message to the two endpoints with the information necessary for establishment of an 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 use 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 must manually increment the parameter bandwidth 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.

From testing, it also seems that only when the work canreinvite a call between the video phone and a desk phone is not activated in the first video. In other cases, to 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

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

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.

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

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

The main changes are:

• Approach completely renovated: the configuration server centralization of services and enabled clients to be notified of calls, as well as the applications to be launched on the 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 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 provide short to load the source code of a special project page on google code. I await your comments, feedback and, above all, a participation in the project!

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

Here I am again to update the pages of this blog asteroid after a period of inactivity, mainly due to the workload of the previous months. During this period I had the opportunity not only to increase experience 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 the 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!

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

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 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 to the agents on the server side
• Support at work after call (PostCall Work) per campaign.

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

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

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, 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/

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

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 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.

Be required:

On the server

• Webserver 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 report, different levels of user access, real-time parsing of the file queue_log, the ability to listen to the recordings of the calls through the streaming.

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

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 has 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 basically two alternative and they are:

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 written in C code and link text2audio.c called to fill in the directory /opt/Loquendo/LTTS7/samples/c/txt2audio .
2. One application dialplan call app_loquendo. This solution, however, does not seem to work with Asterisk 1.4 or Loquendo 7.

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

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 the various changes, this version includes a series of improvements to 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:

• http://downloads.digium.com/pub/telephony/asterisk/ChangeLog-1.4.20-rc2
• http://downloads.digium.com/pub/telephony/asterisk

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