VoIP PBX

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If you are among those who usually install Asterisk from source, available to you here is a handy utility that lets you automatically download the latest stable branch 1.6.
The utility is based on dialog and downloads 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 chosen:

• Asterisk
• asterisk-addons
• libpri
• dahdi

To use Asterisk 1.6 downloaders

~# 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:

• Asterisk Downloader 1.6

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

Recently I had the opportunity to try a SIP videophone unique: it is produced by SVP SBNTech 5000.

The key features of this product are:

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

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

Please note that this "Resubmit" will not happen, despite "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 (W w) / call transfer (to T) / forced hangup (I H).

In all cases, the required bandwidth for a video call is still at least 200kbps, provided that the canreinvite is enabled. In other cases need more bandwidth. Note that, as soon as an active video call, mobile value using bandwidth as low as possible. To get better video quality must increase bandwidth parameter dall'apposito hand panel configuration call.

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

The tests carried out also seems that only when canreinvite operates a call between a mobile phone and the video board is not active in the first video. In other cases, goes hand-selected to not activate the video when you make or accept the call from the video phone.

Very intuitive and simple user interface: all functions are easily accessible thanks to a convenient touch.

The product was kindly made ​​available by www.4geek.it

The manufacturer's page: SBNTech

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

A Few weeks I started from a general Restructuring of Asterisk CTI. The new version, for release in Which is indicatively Planned for early 2009, Represents a Revolution Against The first version of the project.

The main features are:

• Completely renovated Approach: 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, and Will Be Called AstCTIConfigurator 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 | PBX Phone | VoIP | Asterisk CTI | PBX | IP Phones | Networking | Linux

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

The main changes are:

• Approach completely renovated: the centralization of services on the server 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 have a truly multi-platform software.
• Client, called AstCTIClient, rewritten GTK and platform.
• 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 to be loaded at startup AstCTIServer: this makes the server independent of any failures of the MySQL database.
• Multilanguage support.

Of course this is an Open Source project under GPL and a short will load the source code of a special project page on google code. I await your comments, feedback and, above all, a stake in the project!

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

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 not only to increase experience in the field of telephony, but also to make new and important knowledge.

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

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

I expect many of the pages of my blog!

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

Coming soon some improvements in the software AsteriskCTI :

• Support the location with the possibility of simply adding new translations
• Ability to disable user configuration on the client
• Ability to turn off the outbound on the client
• Integrated web browser for a better management of web-applications CTI. The browser has a base level configuration, but several functions can be implemented on request.
• Support for agents on the servers
• Support post-call work (Work PostCall) per campaign.

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

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

The development team has released Asterisk version 1.4.21.

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

Asterisk 1.4.21 is available for download from the Digium site:

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

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

Nicolas Gudin - author of the FOP, the popular Flash Operator Panel - has released a new software designed for Asterisk queue monitoring of statistics called Asternic Callcenter Stats

An excerpt from the home page translated into Italian:

Asternic Call Center Stats is a package for analyzing files queue_log your Asterisk PBX via a web page and view information in real time.

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

Are required:

On the server

• Webserver with PHP support
• MySQL database

On clients

• A browser that supports Javascript
• Flash Plugin

There are two versions: a free licensed with the GPL v3 and a commercial version includes many versions 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.

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

Loquendo TTS is a Text to Speech software that includes commericiale entries for many languages ​​including English, French, German, Spanish, Italian, Portuguese, Greek, Catalan and Swedish.

You RPR has published an interesting voip-info.org wiki page which shows the various techniques of integration of Loquendo with Asterisk.

The methods below are basically two alternatives and they are:

1. A script called loquendo.agi AGI for inclusion 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. An application called app_loquendo dialplan. This solution, however, seems not work with Asterisk 1.4 or 7 with Loquendo.

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

The development team has released version 1.4.20-rc2 Asterisk.

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

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

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 | PBX Phone | VoIP | Asterisk CTI | PBX | IP Phones | Networking | Linux