First Home Automation products are released!

Written by Stéphane LAVAUD on . Posted in Asterisk, IP01, IP02, IP08, VoIP embarqué, VoIP et le web

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.

Skype won’t renew the agreement for “Skype for Asterisk”

Written by Hugo Briand on . Posted in Actualités, Asterisk, VoIP et le web

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

 

[Tutotial] AstCallBack Lite on IP0x

Written by Philippe TRAN BA on . Posted in Asterisk, BR4, IP01, IP02, IP08, Non classé, Technologie mobile, Telephone IP, VoIP embarqué

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)

Login page

    1. Go to Admin > File Editor.

File Editor

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

sip.conf

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

Add context
then, click on Add.
Add

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

Gray bar

    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.

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.
Click here

  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.


[Tutorial] Set up a trunk between two asterisk server

Written by Philippe TRAN BA on . Posted in Asterisk, BR4, IP01, IP02, IP08, Non classé, Routage dans reseau SIP, Telephone IP, VoIP embarqué

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)

Login page

    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)


[Tutorial] “Echo Canceller” Module

Written by Philippe TRAN BA on . Posted in Asterisk, BR4, IP01, IP02, IP08, Non classé, Telephone IP, VoIP embarqué

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.

Switchvoice about to start with new home automation line of products

Written by Philippe TRAN BA on . Posted in Non classé

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

New Pr1 Appliance firmware release + SD card for free

Written by admin on . Posted in Non classé

New firmware (uImage-pr1-v401.img) for PR1 Appliance has been released.

In this release the SD card support is improved.
As an extra benefit for our customers all PR1-Appliance units from now on will be equipped with 2GB (or bigger) SD card plunged in.

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

PR1 Appliance – firmware update

Written by Philippe TRAN BA on . Posted in Non classé

New firmware (uImage-pr1-v400M.img) for PR1 Appliance has been released.

This release is equipped with the latest Switchfin FAX stack.
Now you can enable the  LEC-64-PR1 module from the GUI. For more details please check the FAQ item Why my FXS/FXO/GSM/LEC is not detected in Asterisk

It is always a good idea to BACKUP your configuration files and custom voice prompts (if any) before upgrading!

ZigBee extension to the IP0x PBX series (coming soon)

Written by Philippe TRAN BA on . Posted in Non classé

Have you ever wanted to control your front door lock or your air conditioner using your home PBX system?
Or probably you want to control remotely any other appliance you have at home?
The new ZigBee extension to the IP0x family of PBXs is just for that.
For the control you may use simple call, IVR or SMS.
ZigBee wireless technology makes deployment of the system very easy.

We are going to release the hardware pretty soon.
Please stay tuned

 

IP02/IP04/IP08 firmware update

Written by Stéphane LAVAUD on . Posted in Asterisk, IP02, IP08, VoIP embarqué

New firmware (uImage-ip02-ip08-v383.img) for IP02/IP04/IP08 has been released.

This release you will find:

  • improved FAX stack
  • deny/permit options in the GUI for better PBX security
  • Improved FXS/FXO/SIP status indications in the GUI
  • DynDNS options implemented in the GUI

It is always a good idea to BACKUP your configuration files and custom voice prompts (if any) before upgrading !

Warning for PBX users!

Written by Stéphane LAVAUD on . Posted in Asterisk, BR4, IP01, IP02, IP08, Non classé, PR1, VoIP embarqué, VoIP et le web

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.

IP01/IP01p major release

Written by Philippe TRAN BA on . Posted in Non classé

New firmware (uImage-ip01-v358.img) for IP01/IP01p has been released.

This is a major release so we recommend it. It is equipped with GUI 4.0 which is going to become our default GUI.
Keep in mind that the users manual from the switchvoice site describe GUI 3.0. We will update them soon.

In this release fax2email is not based on Spandsp but Attrafax

From this release we are changing the revision numbering. The new numbering represents directly Switchfin SVN
revisions so it easier to be tracked.
It is always a good idea to BACKUP your configuration files and custom voice prompts (if any) before upgrading !

GSM1 support in the latest Switchfin software

Written by Stéphane LAVAUD on . Posted in VoIP embarqué

Improved version of GSM1 firmware is put in the latest Switchfin software.

Switchvoice customers now can rely on GSM connectivity and at the same time having the latest Switchfin additions available.
IP02/IP04/IP08 firmware supporting the GSM1 update can be found at uImage-ip02-ip08-v370gsm.img
This version needs pin number protected SIM card and you have to set the pin in /etc/init.d/dahd manually like this:

start)  modprobe wcfxs SIM_pin=”1234″ opermode=$opermode;

For inquiries please reffer to the product section GSM1.

IP01 major release

Written by Philippe TRAN BA on . Posted in Non classé

New firmware (uImage-ip01-v358.img) for IP01/IP01p has been released.

This is a major release so we recommend it. It is equipped with GUI 4.0 which is going to become our default GUI.
Keep in mind that the users manual from the switchvoice site describe GUI 3.0. We will update them soon.

In this release fax2email is not based on Spandsp but Attrafax

From this release we are changing the revision numbering. The new numbering represents directly Switchfin SVN
revisions so it easier to be tracked.
It is always a good idea to BACKUP your configuration files and custom voice prompts (if any) before upgrading !