Hi,

welcome to our brand new blog.

After some crash issues, we will move forward to new adventures to present you on a regular basis the last news around video, voice over IP technologies but also the last applications for mobile and web.

Have a pleasent  visite.

Neotiq web crew.

Our first products for home automation have been released.
Now you can control any appliance at home securely using phone call or DTMF/IVR to your home PBX.

Products specifications:
1. WireLess Tele Controller (WLTC). Wireless module for IP01/IP02/IP04/IP08 PBX.
It allows remote devices to be controlled by the PBX

• ZigBee
• WLTC can coexist with FXS/FXO modules
• IVR controllable appliance
• Build in internal antenna
• Covers 200m without obstacles to the remote wireless device
  or 3 concrete walls. (Adjustable power at the antenna)
• Power saving MAC protocol.
• 6LowPAN enabled.

2. WireLess Remote Relay (WLRR). Remote wireless enabled unit able to control single low or high power appliance.

• ZigBee
• Mains pluggable, No installation required
• IVR controllable appliance
• Provides controllable mains socket (250VAC/16A)
• Build in internal antenna
• Covers 200m without obstacles to the WLTC
  or 3 concrete walls. (Adjustable power at the antenna)
• Power saving MAC protocol.
• 6LowPAN enabled.

Digium warned users that Skype won’t renew the license agreement for “Skype for Asterisk”. Skype for Asterisk is a solution to interconnect Skype and Asterisk, both ways. With this interconnector you can register an enterprise Skype account as an Asterisk’s trunk, and to use Skype’s software to register as an Asterisk’s client.

This non-renewal means that the product won’t be sold or activated after July, 26. Exsiting users won’t be affected as Skype insured that support and maintenance of Skype for Asterisk will go on at least until July, 26 of year 2013.

Shall we see the shadow of Microsoft, the new owner of Skype, behind this decision ?

The Digium’s press release :

Skype for Asterisk will not be available for sale or activation after July 26, 2011.

Skype for Asterisk was developed by Digium in cooperation with Skype. It includes proprietary software from Skype that allows Asterisk to join the Skype network as a native client. Skype has decided not to renew the agreement that permits us to package this proprietary software. Therefore Skype for Asterisk sales and activations will cease on July 26, 2011.

This change should not affect any existing users of Skype for Asterisk. Representatives of Skype have assured us that they will continue to support and maintain the Skype for Asterisk software for a period of two years thereafter, as specified in the agreement with Digium. We expect that users of Skype for Asterisk will be able to continue using their Asterisk systems on the Skype network until at least July 26, 2013. Skype may extend this at their discretion.

Skype for Asterisk remains for sale and activation until July 26, 2011. Please complete any purchases and activations before that date.

Thank you for your business.

Digium Product Management

1 – Principle of Astcallback

AstCallBack is a unique solution for mobile phone based on the open-source telephony system of Asterisk.

With AstCallBack, it becomes possible to cut your mobile phone bills. The principle is to use the callback feature of your Asterisk PBX: you choose a person to call, your PBX calls you back then calls your contact and put you together.

This solution is especially advantageous that some numbers are very expensive from your mobile (special number, international) and these calls from abroad (roaming) are sometimes exorbitant.

The “lite” version offers the basic services through a simple and free solution. It is now available on AppStore.

2 – Astcallback settings

1. Connect the PBX to the sector and to the computer with a RJ45 cable and switch it on.
2. 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)
3. Type your login and password (Defaut admin and switchfin)

1. Go to Admin > File Editor.

1. Edit the sip.conf file in the menu on the right.

1. Create the [outpeer] extension by clicking on Add context

then, click on Add.

1. Open the extension by clicking on the “+” icon next to the context. A gray bar appears, click on it.

1. Type in the following text :

type=peer
qualify=yes
fromdomain="your SIP provider address"
fromuser="your SIP Account's username"
username="your SIP Account's username"
secret="your SIP Account's password"
host="your SIP Account's proxy server"
realm="your SIP Account's realm"
insecure=very
nat=yes
canreinvite=no

1. Save.
2. Then, open the file extensions.conf.

1. Create the extension [callback_with_number] like you did for [outpeer]
2. Open it and type in the following text and adapt the first line to your configuration :

exten => _.,1, Dial(ZAP/g3/${EXTEN})
exten => _.,2,Hangup

Note : It is possible that the extension is not fully visible. Click on what exceeds like below.

1. Save.

3 – Caution

BEWARE through your Asterisk server and its AMI interface are available from your iPhone, you also put it accessible to all. It is YOUR RESPONSIBILITY to secure your configuration so that your Asterisk server can not be used by unauthorized persons. The application AstCallBack Pro will support this security.

1 – What is it ?

A SIP trunk is a connection made between 2 SIP servers to allow SIP calls. We want this connection to allow phones from a server A to call a server B. In this article we will explain step by step how to configure a SIP trunk between two Asterisk servers.

2 – Setting up a SIP trunk

Before setting SIP trunks, we need to connect and create extensions. Two for the user’s with the phone A and phone B and two others to identify the trunks A2B and B2A.
1. Connect the PBX to the sector and to the computer with a RJ45 cable and switch it on.
2. 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)
3. Type your login and password (Defaut admin and switchfin)

1. Go to System Setup > Configure trunk

1. Click on VOIP Trunks then on New SIP/IAX Trunk
2. Create a Trunk for transfers from A to B by entering the IP of B and the future A extension on B you will put later. In our example, the B IP is :192.168.1.88 and the extension of A will be : 6222.

1. Go to PBX Features > Incoming rules

1. Create the incoming calls rule. Here, we indicate that we will have to dial the 01 before the normal internal extension.

1. Then, enter the PBX Features > Outgoing rules menu

1. Create a suitable rule. Here it is shown that 2-digit number will be removed (in this case 01).

1. Then, enter PBX Features > Dial Plans

1. Create a dialplan including the rule created previously.

1. Go to Extensions
2. Create a SIP account for the trunk B2A. You should also create an account in order to test the configuration.
3. In order to avoid IP conflicts go Admin > Networks settings and change the IP. Restart.

1. Connect to Server B and make the same manipulations

• Creation trunk B2A
• Creating rules for outgoing calls and incoming calls (which are not necessarily the same as A)
• Creating Dial Plan
• Create the extension for A on B and user.
• Changing the IP address (Optional. Make sure to be in the same subnet as A)

New firmware (uImage-ip02-ip08-v509gsm.img) for IP02/IP04/IP08 has been released.
It is always a good idea to BACKUP your configuration files and custom voice prompts (if any) before upgrading !

New firmware (uImage-ip02-ip08-v383M.img) for IP02/IP04/IP08 has been released.

This release you will find:

• Support of dm9000B Ethernet chips added

It is always a good idea to BACKUP your configuration files and custom voice prompts (if any) before upgrading!
Firmware you can find at the download section of the site

1 – Principe de l’annulation d’Echo

L’annulation d’écho consiste à éliminer l’écho de la diffusion vocale de façon à améliorer la qualité de l’appel. L’annulation de l’écho est souvent nécessaire car les techniques de compression de la voix et de traitement de paquet de délais génèrent une résonance. Non seulement l’annulation de l’écho améliore la qualité de la communication mais elle réduit aussi l’usage de bande passante grâce à sa technique de suppression de silence.

2 – Installation de la carte d’annulateur d’écho

1. Ouvrez le boîtier soigneusement.
2. Posez méticuleusement la carte sur l’emplacement adéquat et enfoncez de manière uniforme.

1. Refermez le boîtier.
2. Connectez le PBX au secteur et à l’ordinateur avec un câble RJ45 et allumez le.
3. Entrez l’IP du PBX dans votre navigateur web (Par défaut : 192.168.1.100 Veillez à être dans le même sous-réseau. Par exemple, prenez comme adresse 192.168.1.2).
4. Entrez votre identifiant et mot de passe (Par défaut admin et switchfin).

1. Allez dans les menus System Setup > Configure Hardware.
2. Vérifiez la présence de la colonne Echo Cancellation dans le tableau car par défaut, le module est actif sur tous les ports compatibles. Ne suivez les étapes suivantes que dans le cas où la carte ne se serait pas activée d’elle même.

1. La carte ne s’est pas activée. Vous devez à présent vous connecter sur le Linux de l’appareil. Pour cela, utilisez un logiciel tel que Putty
   (http://www.chiark.greenend.org.uk/~sgtatham/putty/download.html)
   Ou SecureCRT
   (http://www.vandyke.com/products/securecrt/index.html)
   De cette façon, vous pouvez accéder directement au Linux de votre PBX et utiliser les options avancées en entrant son IP.
   L’utilisateur du SSH est root, le mot de passe est uClinux (La casse est comprise et le mot de passe n’est pas affiché).
2. Faites un :
   vi /etc/asterisk/misdn.conf
3. Allez à la ligne contenant la variable :
   echocancel = 0
4. Changez le 0 en 64, sauvegardez, et quittez.
5. Redémarrez le PBX. A présent, la carte devrait être active. Si ce n’est toujours pas le cas ou si vous rencontrez d’autre problèmes, nous vous invitons à laisser un commentaire sur cet article.

3 – Activer/Désactiver le module à partir du GUI

1. Pour activer ou désactiver le module sur des ports en particulier, il faut vous connecter au GUI (Refaire les étapes de  2.4 à 2.8)
2. Cliquez sur le bouton Edit correspondant aux ports souhaités.

1. Pour activer ou désactiver l’écho cancellation sur un port, il suffit de cocher ou décocher la case en face du port.

New firmware (uImage-br4-v424M.img) for BR4 Appliance has been released.

In this release the LEC support has been improved

Firmware you can get from the link above.

Last few months we have been busy with our first products targeting home automation from the PBX.
We have utilize ZigBee based wireless technology to make the deployment of the products as simple as possible.

In the very near future we will have the following four products ready.

1. Wireless Tele Controller (WLTC). Module in IP01/IP02/IP08. It adds ZigBee connectivity to those PBXs. The module covers wireless connectivity in a typical home environment. (50m and few walls.)

• Seamless integration with the existing PBX feature.  
• 6LowPAN enabled.
• Low cost

2. Wireless Remote Module (WLRM08). Remote ZigBee enabled unit able to sense and control several home appliances.

• Two normally opened relay outputs (250VAC /5A
• One normally closed relay outputs (250VAC /5A)
• One switching relay output (250VAC /5A)
• Three digital inputs
• 6LowPAN enabled.
• Relays control thru IVR or SMS.
• Web based status information of the relays outputs and digital inputs
• SMS or call notifications on digital input change.
• Low cost

3. Wireless Remote Relay (WLRR). Remote wireless enabled unit able to control single low or high power appliance.

• Mains pluggable, No installation required
• Provides controllable mains socket (250VAC/16A)
• Web based status information of the relay output
• 6LowPAN enabled.
• Low cost

4. Wireless Remote Relay and Thermometer (WLRRt). Remote wireless enabled unit able to control single low or high power appliance. The unit is able to measure the environment temperature

• Mains pluggable, No installation required
• Provides controllable mains socket (250VAC/16A)
• Web based status information of the relay output
• Web based information for the measured temperature.
• SMS or call notifications related with the temperature information.
• 6LowPAN enabled.
• Low cost