VoIP PBX

If you are among those who usually install Asterisk from source, here's to you with a handy utility that lets you automatically download the latest stable branch 1.6.
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 chosen:

• 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:

• Asterisk Downloader 1.6

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

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Development of IVR systems, call center, VoIP PBX.

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Lately, I've got to try a SIP videophone unique: it is produced by 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 sources. 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 to avoid, as far as possible a configuration in which Asterisk should worry about 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" allows in fact to Asterisk, once that the call has been accepted, to send a new message INVITE to the two endpoints with the information necessary to the establishment of a audio / video communication direct between the same.

Please note that this "reINVITE" will not happen, despite "canreinvite = yes", if:

1) one of 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 (W w) / call transfer (to T) / forced hangup (I H).

In all cases, the band required for a video call is however of at least 200kbps, always that the canreinvite is active. In other cases need more bandwidth. Attention to the fact that, as soon as active a video call, the phones use a value of bandwidth as low as possible. To get better video quality must be increased by hand dall'apposito bandwidth parameter in the configuration panel called.

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 banda request.

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

Very intuitive and simple user interface: all functions are easily accessible with the convenient touch screen.

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

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Development of IVR systems, call center, VoIP PBX.

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