Désolé, cet article est seulement disponible en English.

1 – Principe d’Astcallback

AstCallBack est une solution unique pour téléphone mobile se basant sur le système de téléphonie open-source Asterisk.

Grâce à AstCallBack, il devient possible de réduire vos factures de téléphonie mobile. Le principe consiste à utiliser la fonctionnalité de rappel de votre PABX Asterisk: vous choisissez une personne à appeler, votre PABX vous rappelle puis appelle votre destinataire et il vous mets en relation.

1 – Qu’est-ce qu’un trunk SIP ?

Un trunk SIP est une connexion SIP faite entre 2 serveurs SIP pour faire passer des appels. On veut que cette connexion permette à des téléphones d’un serveur A d’appeler des téléphones d’un serveur B. Dans cet article, nous allons expliquer pas à pas comment configurer un trunk SIP entre 2 serveurs Asterisk.

1 – Principle of echo cancellation

Echo cancellation is the proceed to remove the echo of the voice broadcasting so as to improve the quality of the call. The echo cancellation is often needed because the compression techniques of voice and packet processing delays generate a resonance. Echo cancellation does not only improve call quality but it also reduces bandwidth usage due to its suppression technique of silence.

2 – Echo cancellation setting

1. Open the case carefully.
2. Meticulously put the card into the right location and press evenly.

1. Close the box.
2. Connect the PBX to the sector and to the computer with a RJ45 cable and switch it on.
3. Enter the PBX IP in your web browser (Default : 192.168.1.100 Make sure that you are in the same subnet. For instance :192.168.1.2)
4. Type your login and password (Defaut admin and switchfin)

1. Go to System Setup > Configure Hardware
2. Check for column Echo Cancellation in the table because by default, the module is active on all compatibles ports. Do the following steps only in the case that the card would not be activated by itself.

1. The card was not activated. You must now sign on the Linux machine. For this, use a program like Putty
   (http://www.chiark.greenend.org.uk/~sgtatham/putty/download.html)
   Or SecureCRT
   (http://www.vandyke.com/products/securecrt/index.html)
   This way, you can go directly to your PBX Linux and use advanced options by entering its IP.
   The user of SSH is root, and the password is uClinux (The password is not displayed and is case-sensitive).
2. Type :
   vi /etc/asterisk/misdn.conf
3. Go to the line containing the variable :
   echocancel = 0
4. Change the 0 in 64, save and exit.
5. Restart the PBX. Now the card should be active. If this is not the case or if you encounter other problems, please, fell free to bother us and put a comment.

3 – Enable/Disable the module from the GUI

1. To turn on or off the module on specific ports, you must connect to the GUI (Repeat steps 2.4 to 2.8)
2. Click on the Edit button corresponding to the desired ports.

1. To enable or disable echo cancellation on a port, just check or uncheck the box in front of the port.

Un nouveau firmware (uImage-ip02-ip08-v475gsm.img) pour IP02/IP04/IP08 est maintenant disponible.

Cette nouvelle version inclue:

• le support des SMS via le module GSM1
• Un serveur HTTP plus rapide

Pensez à faire un “backup” avant de modifier le firmware.

Un nouveau firmware (uImage-ip02-ip08-v383.img) pour IP02/IP08 vient d’être mis en ligne.

Dans cette nouvelle version, vous trouverez:

• amélioration de la pile FAX
• Option deny/permit sur la GUI pour une meilleure sécurisation du PBX
• Amélioration du status FXS/FXO/SIP dans la GUI
• Option DynDNS implémenté dans la GUI

Nous avons déjà publié quelques informations pour se protéger avec Asterisk. Voici un communiqué de Switchvoice qui reprend les même bases.

Les anglophobes s’abstiendront…

******

Recently few clients reported that their PBXs have been hacked.

In case any of you were wondering why there has been a fairly notable upswing in the attacks happening on SIP endpoints,
the answer is “script kiddies.”  In the last few months, a number of new tools have made it easy for knuckle-draggers to attack
and defraud SIP endpoints including Asterisk-based systems as the one Switchvoice manufacture.
There are easily-available tools that scan networks looking for SIP hosts, and then scan hosts looking for valid extensions,
and then scan valid extensions looking for passwords.

There are few simple things you may do to increase the security of your PBXs.

1. Put your PBX behind router/firewall and open given port only if necessary.
2. Use not trivial SIP/IAX user names and long difficult passwords. Never use user name and password being the same.
3. Use the “permit=” and “deny=” lines in sip.conf to only allow a reasonable subset of IP addresses to reach each listed extension/user in your sip.conf file.
   As general practice always do this in case you need to connect to your PBX from outside of your local network and
   therefore you need to open SIP port 5060 on your router.
   This last option we plan to put in the GUI with the next release.
4. You may consider changing the SSH password of your PBXs being more complex.

Please pay attention before you get hacked.
After all VOIP is to make the communication easier, more convenient and cheap.