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Sunday, 30 September 2012

Featured: AOKP Jelly Bean Build 4 released! Adds Samsung Galaxy S3, AT&T One X and more Read more at http://androidheadlines.com/2012/09/featured-aokp-jelly-bean-build-4-released-adds-samsung-galaxy-s3-att-one-x-and-more

Here we go! If you’re a crackflasher, it’s time to go ahead and flash the newest build of AOKP. With a new build comes a few new features, and a few new devices added to the list of supported for Jelly Bean builds. The new devices include
  • Samsung Galaxy S3 – US Cellular (d2usc)
  • AT&T HTC One X (evita)
  • ASUS Transformer (tf101)
Nice to see the original Transformer getting some Jelly Bean love from AOKP. There is also a few more team members that have joined AOKP. Welcome the very talented BMc80GT, K900 and rohan. And now for the new features in Build 4:
  • Custom NavBar Ring on tablets
  • Force dual panel view on any device
  • Flip or shake your device to snooze/dismiss the alarm (check alarm settings)
  • Take screenshot from NavBar
  • Enabled USB mass storage emulation for certain devices (i9*)
The team has also fixed the NavBar custom icon bug found in previous builds. As well as some minor issues with Auto Brightness and fastcharge.
Before flashing, please BACK UP! As noted in the release on AOKP’s website, on certain devices this new build will wipe your device. You should always backup before flashing a new ROM anyways, but it still needs to be said.
So let’s get to flashing. I’ll be flashing AOKP Build 4 onto my Galaxy Nexus in a few minutes. Who else is flashing tonight as well?

Asterisk Application: IP PBX, Part - 3

Step 4: Configure Connections

Now that Asterisk is installed and running you need to edit the system configuration files to implement connections to VoIP and PSTN services. Since this step is common to all applications (Asterisk doesn't do much good if it is not connected to anything) it contains information on creating both service connections (connections to VoIP or PSTN services) and endpoint connections (connections to phones or terminal adapters). Some applications require both service and endpoint connections (PBX, ACD) while others may require only service connections.

DAHDi Connections

DAHDi is the "Digium Asterisk Hardware Device Interface" project and is the standard means for connecting Asterisk with PSTN interface cards. If your application doesn't require PSTN connections (i.e. your solution is VoIP-only) you can skip to the VoIP Connections section.
To configure your DAHDi interfaces, be sure that your cards are installed and properly connected to the computer. If you are using analog cards with FXS (station) ports, remember to connect one of the power connectors from the PC's power supply with the molex coupler on the edge of the card. FXS draws more power than the PCI and PCI Express buses can provide, so the connection is mandatory. Analog with FXO (line) interfaces only don't need the power connection.
To configure the cards you need to edit two files. This can be done using any standard text editor (emacs, vi, vim, gedit, kedit, etc.). Note that you will need to be logged in as the root user to edit these files (or use the sudo command if your Linux distribution uses sudo instead of direct root access). The first file is the /etc/dahdi/system.conf file. This file configures the parameters for the low-level card drivers. There are only a few options that need to be set in this file. Below are configurations for both a system with one digital card (in T1 mode):
# Autogenerated by /usr/sbin/dahdi_genconf on Wed Sep 23 08:25:39 2009
# If you edit this file and execute /usr/sbin/dahdi_genconf again,
# your manual changes will be LOST.
# Dahdi Configuration File
# This file is parsed by the Dahdi Configurator, dahdi_cfg
# Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) 
# termtype: te

# Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2" 
# termtype: te

# Global data

loadzone        = us
defaultzone     = us

VoIP Connections

Over the next few days we will be filling in the remainder of these tutorials. Sorry for any inconvenience. Please check back soon.

Asterisk Application: IP PBX, Part - 2

Step 2: Select Your Computer Hardware

ComputerAsterisk can run on virtually any modern computer, but when building a production telephony application server you should follow a few basic best-practice guidelines. Click the "More" link below to learn the basic requirements for a solid Asterisk server.
Telephony systems are generally mission-critical components and therefore need to be as reliable as possible. So while you can build an Asterisk server using a Start with a reasonably powerful and reliable server-class system as your platform. While not required, it's generally a good idea to use a system with redundancy features including mirrored hard disk drives (RAID 1, 5 or 10) and dual power supplies.
Cooling is also an important issue. PSTN interface cards can add to the overall heat load of the server. Failure to provide proper ventillation can cause stability issues and can lead to premature failure of critical components.
CPU and memory requirements vary depending on the application. Small systems can be built on embedded processors using only a few megabytes of memory. Large scale system that process thousands of simultaneous calls require significantly more horsepower and memory. The safest bet is to go with the most powerful CPU and the most memory that fits within your budget.
If your application calls for TDM hardware, be sure that your server includes the correct type of card slot. Cards are available in PCI and PCI Express form factors and at a variety of voltages. See the "Select Your Telephony Hardware" section for more information on the various interconnect formats.

Step 3: Install Linux & Asterisk

Once you have your Asterisk hardware the next step is software. You will either need to install Linux or use a ready-to-run distribution to install Linux, Asterisk and various related software packages. Since these application tutorials are intended to help you create custom telephony applications we will start with a generic installation of CentOS 5.3 and then install Asterisk from the Yum repository. This make it relatively easy to keep Asterisk up to date and avoids the complexities of hand compiling the Asterisk source code.

Download CentOS Linux 5.3

The first step is to download a copy of the CentOS 5.3 installation image. There are two supported system architectures available, a 32-bit version of Linux and can run on either 32-bit or 64-bit systems, and a 64-bit version that runs only on x86_64 hardware. If in doubt, download the 32-bit version. Most CentOS mirrors offer the distribution as either 5 CD images or a single DVD image. Either method works for the purposes of this tutorial. Use the CentOS 5 mirrors list to select a mirror. Most mirrors host the CD images and a BitTorrent seed file for downloading the DVD image. To download the DVD image use a BitTorrent client.
Download or copy the .iso image to a computer with a CD or DVD burner (writer). Keep in mind that the images are roughly 700 MB each and the DVD image is over 3 GB.

Burn The CentOS .iso To CDs or DVD

Use your CD or DVD burner software to burn the ISO image to an actual CD. Note that if you are installing on virtual machine you can generally use the ISO image without burning it to physical media.

Install CentOS

To install CentOS Linux, insert the newly burned CD in the CD or DVD drive of the target computer and boot. Be sure that your system is set to boot from the CD or DVD drive. (You can either adjust the boot order in the system BIOS or use a one-time boot menu if your system supports it.)
Full details of the CentOS installation are beyond the scope of this tutorial. Several excellent quick-start tutorials can be found at Howto Forge, including this one. For more detailed installation instructions you can also refer to the CentOS Installation Guide . In general it is safe to select the default options throughout the installation process. Be sure to select a secure root password when prompted. IP Telephony systems are a frequent target for hackers and maintaining system security is extremely important.
Note that the installation of graphical environments (Gnome, KDE, etc.) is perfectly acceptable on systems used for development or unit testing. When building production servers or systems designed for load testing it is recommended that the graphical interfaces and subsystems (frame buffers) be omitted.

Add Asterisk Yum Repositories

After the install is complete and your new CentOS Linux system is up and running, log in as the root user with the password you set during the installation. (Note that you generally want to avoid logging in as root. Create a non-privileged user account for day to day operations.)
Now, use the text editor of your choice to create a new file named "centos-asterisk.repo" in the "/etc/yum.repos.d" folder. Add the following text to the file:
name=CentOS-$releasever - Asterisk - Tested

name=CentOS-$releasever - Asterisk - Current
Save the new file and create another named "centos-digium.repo" and insert the following text:
name=CentOS-$releasever - Digium - Tested

name=CentOS-$releasever - Digium - Current
Now you should be ready to install Asterisk. To start the installation, open a terminal window and type the following:
[root@localhost~]# yum install asterisk16 asterisk16-configs asterisk16-voicemail dahdi-linux dahdi-tools libpri
The system will connect with the Asterisk and Digium yum servers, download the necessary packages for Asterisk 1.6 and install them. For a detailed view of sample output from the install see the Yum installation page.
When the installation is complete, reboot your system to activate Asterisk and DAHDi. By default DAHDi will start automatically. To enable auto-start of Asterisk, run the following command:
[root@localhost~]# chkconfig asterisk on
To manually start Asterisk you can use the following:
[root@localhost~]# service asterisk start
To stop Asterisk, execute:
[root@localhost~]# service asterisk stop

Test Your Asterisk Installation

Once the Asterisk service is started you should be able to access the Asterisk command line interface from the Linux command line as follows:
[root@localhost~]# asterisk -r
The system should respond with something similar to:
Asterisk, Copyright (C) 1999 - 2009 Digium, Inc. and others.
Created by Mark Spencer 
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
Connected to Asterisk currently running on localhost (pid = 3052)
Verbosity is at least 3
Congratulations, you now have Asterisk installed and running. To exit from the Asterisk CLI, simply type 'exit'.

Asterisk Application: IP PBX, Part - 1

sterisk was originally built as a PBX and today represents an astonishing 18% of global market for business telephone systems. The base feature set includes many of the most popular and powerful PBX functions.
Tapping the power of Asterisk requires some knowledge of Linux, telephony, basic script programming and IP networking. For those who would rather point and click than compile and script, Digium offers Switchvox, a complete IP PBX system based on Asterisk.
VoIP Gateway

Step 1: Select Your Telephony Hardware

Telephony Card Asterisk applications that connect with legacy telephony systems (PBXs or the PSTN) require telephony interface hardware. Small system generally use analog or ISDN BRI connections. Larger systems (more than 12 lines) frequently use T1, E1 or J1 digital connections. If you're new to telephony, check out the Asterisk telephony by clicking the "More" link below.
Note that not every Asterisk implementation requires telephony hardware. Systems that are connected only by VoIP connections communicate using the host computer's Ethernet port.

Connection Types

Analog connections are commonly used in small businesses and homes. Each analog connection uses a single pair of copper wires. Asterisk connects with analog lines using an analog interface card that converts the voice and signaling information into Asterisk's native digital format.
ISDN BRI connections are digital telephone lines that have replaced analog lines in some places. BRI is very popular in Germany and is also common in businesses in the UK. BRI connections can carry up to two conversations at the same time and support some advanced features not available with analog connections. ISDN can use either a two-wire "U" interface or a four-wire "S/T" interface.
T1, E1 and J1 are standards for high capacity telephony connections. T1 "trunks" or "spans" are the standard in the United States. A T1 can carry as many as 24 simultaneous conversations. E1 is popular throughout much of the rest of the world. E1s are slightly higher bandwidth and can carry up to 30 simultaneous calls. J1 trunks are essentially the Japanese version of the US T1 standard.
There are several different services offered over T1, E1 and J1 connections. The most popular service type is ISDN PRI. PRI circuits use what is known as "out-of-band" signaling -- that is, one of the 24 channels (T1) or two of the 32 channels (E1) is reserved for sending call management messages.
T1/E1/J1 lines can also be used as data carriers for providing Internet or private data network services using the HDLC protocol.
For more information on both BRI and PRI forms of ISDN, check out the Wikipedia article.

Interface Hardware

Asterisk connects with analog and digital telephony connections through either a gateway card that is installed in the host computer (the computer running Asterisk) or through an external gateway device. Internal gateway cards generally connect through the computer's internal expansion bus. Cards are available in "PCI" and "PCI Express" (or PCIe) form factors. External gateways connect with Asterisk over the local area network (LAN) or the PC's USB bus.
Internal gateway cards are the most common means of connecting Asterisk to the PSTN or to a legacy telecom system. Cards typically fall into the same categories as telephony connections: analog, ISDN-BRI, and T1/E1/J1 devices. (There are a few hybrid devices on the market that support both analog and ISDN-BRI.)

Analog Cards

Analog cards are available in various capacities, ranging from a single port (which connects a single analog telephone line and thus a single telephone call) up to a maximum 24 ports.
TDM800Low density analog cards generally use the same kind of connector as most home and small business phone devices: the RJ-11 jack. Each jack on the back of the card takes a single RJ-11 phone cable. The other end of the cable plugs into a telco phone jack or an analog port on the legacy PBX.
Note in the image to the left that the card (an 8 port model made by Digium) the red blocks to the right-hand side of the card. These are daughter modules (small circuit cards that attach to the main card) that determine the function of each of the ports on the card. Analog ports can either connect to an analog line from the telephone company using a port connected to an "FXO" module, or can power and control an analog phone using an "FXS" module.
Most analog card manufacturers build analog cards with interchangeable modules. By including both FXO and FXS modules, a card can offer both FXO (line) and FXS (phone) capabilities. This makes it simple to build an Asterisk-based application that can both connect to the PSTN and control analog devices like fax machines, credit card terminals or TDDs.
TDM2400High density analog cards often use an RJ-21 (or "amphenol") connector and require an opposite gender RJ-21 connection from the telco or PBX. Note the green modules on the card in the image to the left. This 24 port card has been configured with three four-port FXS modules (green) and three four-port FXO modules (red). This allows it to connect 12 analog devices and 12 analog phone lines.
FXS Ports In Analog Gateways
VoIP gateways built to connect legacy equipment (PBXs, key systems) with VoIP services generally use FXS ports. FXS ports provide dial tone and line voltage to a phone, exactly like the phone company's line does. This means that FXS ports can be connected to "line" or "trunk" ports on the legacy system and can emulate telco analog lines.
FXO Ports In Analog Gateways
Using FXO (line) ports to connect analog PBX ports with VoIP phones or remote VoIP servers is another common Asterisk gateway application. In this scenario, analog station ports on the PBX are connected to FXO ports on the Asterisk gateway card. When the PBX sends a call to the cross-connected analog station port, Asterisk forwards it as a VoIP call to the designated endpoint.
The limitation of this arrangement is the one-extension/one-port nature of PBX analog stations. The signaling capabilities of analog station ports are generally very limited. This means that the port can only respond to calls to the single extension number with which it is associated. It therefore cannot be used as a shared connection between the PBX and the Asterisk (and whatever connects with the Asterisk).

Digital Cards

TE412pDigital cards allow Asterisk to connect with T1, E1 and J1 digital lines (sometimes called "trunks"). Digital cards include one or more ports, each of which connects to an individual digital circuit. Trunks are often referred to as "spans", thus a single port card is a "single span" device, while a four port card is a "quad span" device. Most digital cards connect using RJ-45 jacks (the same kind of jack as is commonly used for Ethernet connections). Connections between the card and a telephone company T1 line are connected using a "straight-through" cable (exactly like the cables that connect Ethernet ports). Connections between the card and a local PBX or other "CPE" device require a "cross-over" cable or a straight-through cable with a cross-over adapter.

Asterisk Applications

Asterisk is named after the Unix/Linux wild-card character because, like the wild-card Asterisk can be virtually anything. This section covers a number of basic examples and provides recipes for creating some of the more common configurations. We are always looking for additional recipes, white papers and case studies, so please let us know if you have an example to post.
Asterisk is often referred to as "the open source PBX" and it's true that you can use Asterisk to build a PBX, in fact, we've done that work for you with our line of turnkey Switchvox solutions, but a PBX is only one of many applications you can build with Asterisk.
VoIP Gateway
Gateways connect legacy phone equipment to modern VoIP systems. Asterisk supports many different protocols to bridge the modern and legacy communication worlds together.
Asterisk has an integrated voicemail application that enables developers to to build a world class voicemail system at a fraction of the cost of commercial solutions. Some of the largest voicemail implementations in the world are built using Asterisk.
Interactive Voice Response enables companies to improve customer service, reduce repetitive tasks and save money. Asterisk makes it easy to build IVR applications.
Audio Conferencing
Asterisk-based conference servers represent one of the most compelling solutions. Asterisk can scale up to handle large sophisticated multi-party conference applications or down to manage a conference bridge for intra-company use.
The limits of Asterisk are the limits of your imagination. Here we have outlined a number of the most common solutions built on Asterisk. We invite you to explore the power of Asterisk and to see what you can build.

How does FAX work in VOIP environments?

How does FAX work in VOIP environments?

FAX was designed for analog networks, and does not travel well over a VOIP network. The reason for this is that FAX communication uses the signal in a different way to regular voice communication. When VOIP technologies digitize and compress analog voice communication it is optimized for VOICE and not for FAX. Subsequently, there are a number of things you need to take note of when you move to a VoIP Phone System.
If you want to continue using your old fax machine, and you want to connect to your VoIP phone system, its best to use a VoIP Gateway and an ATA that supports T38. T38 is a protocol designed to allow fax to 'travel' over a VoIP network. An example configuration of such a setup can be found here.
It is also possible to convert to computer based fax and choose a VoIP phone system that supports fax. 3CX Phone System for Windows includes a full featured fax server that is able to receive faxes and forward them in PDF format to e-mail. Faxes can be sent from anywhere in the network using theMicrosoft Fax client & fax server (which comes free with Windows Server 2003 and 2008)
Another way to deal with fax when you switch to voip are to connect the fax machine directly to the existing analog phone line and bypass your VOIP system.

10 reasons to switch to an IP PBX

10 reasons to switch to an IP PBX

The benefits of replacing your old PBX with an IP PBX
What is an IP PBX?An IP PBX is a complete telephony system that provides telephone calls over IP data networks. All conversations are sent as data packets over the network.
The technology includes advanced communication features but also provides a significant dose of worry-free scalability and robustness that all enterprises seek. The IP PBX is also able to connect to traditional PSTN lines via an optional gateway - so upgrading day-to-day business communication to this most advanced voice and data network is a breeze!
Enterprises don’t need to disrupt their current external communication infrastructure and operations. With IP PBX deployed, an enterprise can even keep its regular telephone numbers. This way, the IP PBX switches local calls over the data network inside the enterprise and allows all users to share the same external phone lines.
How it works

Figure 1 - How an IP PBX integrates into the network
An IP PBX or IP Telephone System consists of one or more SIP phones, an IP PBX server and optionally a VOIP Gateway to connect to existing PSTN lines. The IP PBX server functions in a similar manner to a proxy server: SIP clients, being either soft phones or hardware-based phones, register with the IP PBX server, and when they wish to make a call they ask the IP PBX to establish the connection. The IP PBX has a directory of all phones/users and their corresponding SIP address and thus is able to connect an internal call or route an external call via either a VOIP gateway or a VOIP service provider. More information and commonly asked questioned about IP PBXs can be found on IP PBX, SIP & VOIP FAQ

Benefit #1: Much easier to install & configure than a proprietary phone system:

An IP PBX runs as software on a computer and can leverage the advanced processing power of the computer and user interface as well as Windows’ features. Anyone proficient in networking and computers can install and maintain an IP PBX. By contrast a proprietary phone system often requires an installer trained on that particular proprietary system!

Benefit #2: Easier to manage because of web/GUI based configuration interface:

An IP PBX can be managed via a web-based configuration interface or a GUI, allowing you to easily maintain and fine tune your phone system. Proprietary phone systems have difficult-to-use interfaces which are often designed to be used only by the phone technicians.

Benefit #3: Significant cost savings using VOIP providers:

With an IP PBX you can easily use a VOIP service provider for long distance and international calls. The monthly savings are significant. If you have branch offices, you can easily connect phone systems between branches and make free phone calls.

Benefit #4 Eliminate phone wiring!

An IP Telephone system allows you to connect hardware phones directly to a standard computer network port (which it can share with the adjacent computer). Software phones can be installed directly onto the PC. You can now eliminate the phone wiring and make adding or moving of extensions much easier. In new offices you can completely eliminate the extra ports to be used by the office phone system!

Benefit #5: Eliminate vendor lock in!

IP PBXs are based on the open SIP standard. You can now mix and match any SIP hardware or software phone with any SIP-based IP PBX, PSTN Gateway or VOIP provider. In contrast, a proprietary phone system often requires proprietary phones to use advanced features, and proprietary extension modules to add features.

Benefit #6: Scalable

Proprietary systems are easy to outgrow: Adding more phone lines or extensions often requires expensive hardware modules. In some cases you need an entirely new phone system. Not so with an IP PBX: a standard computer can easily handle a large number of phone lines and extensions – just add more phones to your network to expand!

Benefit #7: Better customer service & productivity:

With an IP PBX you can deliver better customer service and better productivity: Since the IP telephone system is now computer-based you can integrate phone functions with business applications. For example: Bring up the customer record of the caller automatically when you receive his/her call, dramatically improving customer service and cutting cost by reducing time spent on each caller. Outbound calls can be placed directly from Outlook, removing the need for the user to type in the phone number.

Benefit #8: Twice the phone system features for half the price!

Since an IP PABX is software-based, it is easier for developers to add and improve feature sets. Most VOIP phone systems come with a rich feature set, including auto attendant, voice mail, ring groups, advanced reporting and more. These options are often very expensive in proprietary systems.

Benefit #9 Allow hot desking & roaming

Hot desking – the process of being able to easily move offices/desks based on the task at hand, has become very popular. Unfortunately traditional PBXs require extensions to be re-patched to the new location. With an IP PBX the user simply takes his phone to his new desk – No patching required!
Users can roam too – if an employee has to work from home, he/she can simply fire up their SIP software phone and are able to answer calls to their extension, just as they would in the office. Calls can be diverted anywhere in the world because of the SIP protocol characteristics!

Benefit #10 Better phone usability: SIP phones are easier to use

Employees often struggle using advanced phone features: Setting up a conference, transferring a call – On an old PBX it all requires instruction.
Not so with an IP PBX – all features are easily performed from a user friendly Windows GUI. In addition, users get a better overview of the status of other extensions and of inbound lines and call queues via the IP PBX Windows client. Proprietary systems often require expensive ‘system’ phones to get an idea what is going on on your phone system. Even then, status information is cryptic at best.
ConclusionInvesting in a software-based IP PBX makes a lot of sense, not only for new companies buying a phone system, but also for companies who already have a PBX. An IP PBX delivers such significant savings in management, maintenance, and ongoing call costs, that upgrading to an IP PBX, should be the obvious choice for any company.

What is T38 fax?

What is T38 fax?

T38 is a protocol that describes how to send a fax over a computer data network. T38 is needed because fax data can not be sent over a computer data network in the same way as voice communication. See How does FAX work in VOIP environments? for more information about this. In essence, with T38 a fax is converted to an image, sent to the other T38 fax device and then converted back to an analog fax signal. Most VoIP Gateways and ATA's now support T38 reliably. An example fax setup using a Linksys 3102 can be found here
T38 is described in RFC 3362, and defines how a device should communicate the fax data. In the picture above both the gateway and the fax machine behind the gateway would have to be capable of T38. For the G3 fax machine on an analog line, this process will be transparent. The analog fax machine does not need to know T38.
3CX Phone System for Windows includes a full featured T38 fax server that allows faxes to be sent and received from anywhere in the network. Faxes can be received as PDF and forwarded via email. Faxes can be sent from anywhere in the network via Microsoft Fax Services (which is included with Windows 2003 and 2008)

What is a STUN Server?

What is a STUN Server?

A STUN (Simple Traversal of User Datagram Protocol [UDP] Through Network Address Translators [NATs]) server allows NAT clients (i.e. computers behind a firewall) to setup phone calls to a VOIP provider hosted outside of the local network.
The STUN server allows clients to find out their public address, the type of NAT they are behind and the internet side port associated by the NAT with a particular local port. This information is used to set up UDP communication between the client and the VOIP provider and so establish a call. The STUN protocol is defined in RFC 3489.
The STUN server is contacted on UDP port 3478, however the server will hint clients to perform tests on alternate IP and port number too (STUN servers have two IP addresses). The RFC states that this port and IP are arbitrary.
More information about STUN and VoIP in general can be found in our SIP / VoIP Video tutorials, 'Voip Nuggets'. VoIP Nuggets are short youtube technical training tutorials about VoIP. Click here for the latest list of VoIP Nuggets
Stun functionality is seamlessly handled by 3CX Phone System for Windows - an easy to install windows based PBX.


STUN is a standardized set of methods, including a network protocol, used in NAT traversal for applications of real-time voice, video, messaging, and other interactive IP communications. STUN is an acronym for Session Traversal Utilities for NAT, and is documented in RFC 5389.[1] RFC 5389 obsoletes the previous specification, entitled Simple Traversal of User Datagram Protocol (UDP) through Network Address Translators (NATs), documented in RFC 3489.[2] The obsolete version of STUN, sometimes referred to as Classic STUN, was intended as a complete solution for NAT traversal, and featured an algorithm to allow endpoints to determine NAT behaviour. The current version of STUN is presented as a tool to be used by other protocols, such as ICE. STUN removes the NAT classification algorithm and defines an extensible packet format.
The STUN protocol allows applications operating behind a network address translator (NAT) to discover the presence of the network address translator and to obtain the mapped (public) IP address (NAT address) and port number that the NAT has allocated for the application's User Datagram Protocol (UDP) connections to remote hosts. The protocol requires assistance from a third-party network server (STUN server) located on the opposing (public) side of the NAT, usually the public Internet. The original version of the protocol also specified methods to ascertain the specific type of NAT, but those methods have been deprecated in the newer specification, because of the plethora of specific NAT implementation behavior in various networking equipment and the resulting intractability of the problem and the deficiencies of the method used.

Classic STUN NAT characterization algorithm

Classic STUN specified an algorithm to characterize NAT behavior according to the address and port mapping behavior. This algorithm is not reliably successful and only applicable to a subset of NAT devices deployed.
The algorithm consists of a series of tests to be performed by an application. When the path through the diagram ends in a red box, UDP communication is not possible and when the path ends in a yellow or green box, communication is possible.
STUN Algorithm3.svg

Everything About ENUM

How does ENUM affect the North American Numbering Plan (NANP)?

ENUM does not change the NANP. In fact, it does not change telephony numbering or its administration in any way. ENUM uses existing numbers, and thus will not drain already scarce numbering resources.

What are SRV and NAPTR Records?

SRV and NAPTR are DNS Resource Records that contain information about resources, services and applications associated with a specific phone number.

What happens if a user dials a number that cannot be resolved by DNS?

Where a number cannot be resolved by DNS, a 404-Not Found error message is returned to the device or program initiating the call. This error message is displayed in the case of a Web browser. With IP telephones, the call will be diverted to the PSTN and connected the traditional way.

What happens if a user dials an emergency number?

Emergency numbers are not considered part of the E.164 and ENUM services, as they are locally designated by each country’s regulatory authority. A person dialing an emergency number with an IP phone will be connected via the PSTN.

What protocol does ENUM use for Internet Telephony?

ENUM itself is "protocol-agnostic" because it is application-agnostic. It does not specify what applications a particular number is associated with, but instead provides a unified way of discovering resources associated with it. It can, for example, work with either H.323 or SIP.

What can be said about the skepticism around VoIP?

VoIP will most certainly become a reality in the near future, where Internet telephony is fully integrated with traditional telephony on a global scale. Although in its early stages, VoIP is always evolving and improving. It should be remembered, however, that ENUM is not intended to facilitate VoIP solely; it is intended to work with a number of applications.

What is SIP? How does ENUM relate to SIP?

Session Initiation Protocol (SIP) is the initiator of interactive communications sessions between users. It also terminates those sessions and modifies sessions. With ENUM, SIP can be used to initiate attempts to multiple locations to find the user receiving the call.

Could ENUM be used to provide telephone number portability?

ENUM is not intended for this use, although it may have a place in countries that do not have a centralized database administration. There are implications and limitations in using ENUM in this way, because ENUM is a shared resource discovery service rather than an industry provisioning service. Countries that deploy number portability have telephone service providers that are generally required to comply with regulatory and industry processes and procedures regardless of the underlying technology they employ for telephony service delivery. It’s important to highlight that the ways in which ENUM will need to be deployed must be consistent with applicable national requirements; it does not create an alternate numbering system with its own set of rules and policies.

How is the user of a number authenticated?

Users could be corporations, individuals, government agencies, military organizations and hosts of other non-individual users. Service providers typically assign large blocks of numbers to these entities; the telecom manager within these entities then assigns numbers to users, so even the service providers cannot identify the users for a large portion of the allocated numbers. This is an unresolved issue, but one that must be resolved prior to deploying a robust and secure ENUM service. It is likely that the service provider that allocated the number(s) to the user will be involved in the process of authentication.

What about private numbering plans within a company?

The ENUM protocol can be used in private numbering plans the same way it can be used in the public E.164 numbering plan. The Internet Telephony gateway or proxy needs some intelligence to "decode" a particular dialing string and then decide how to look up resources for that particular number. Instead of looking for resources in e164.arpa, the gateway or proxy would look for SRV or NAPTR records for private numbers under some other structure, such as e164.bigcompany.com.

Are users going to have to pay to have their telephone numbers ENUM-provisioned?

Probably, but it is likely that the costs will be indirectly recovered through the underlying prices for ENUM-enabled services that subscribers pay. This is a DNS-based system, and someone must pay to have a domain name registered in DNS. Listing telephone numbers will be no different. Whether the cost will be charged directly to the subscriber or will be an indirect charge as part of some larger services will depend on those offering the services.
It is important to remember that a user does not have to have ENUM list their phone number. ENUM would be a subscriber-controlled "opt-in" system to "announce," over the Internet, the availability of a particular telephone number to accept service sessions and how to manage those sessions as a result of having subscribed to an ENUM-enabled service. If a customer does not have an Internet telephony device or service, the associated phone number will likely not be listed. On the other hand, subscribers may not necessarily be aware that they have subscribed to such a service, and that they have had ENUM provisioned for that service by their service provider on their behalf.

Are users going to have control over how this system is used with their phone numbers?

With Public ENUM, yes. To reiterate, the first principle in the creation and operation of a global ENUM service is that phone number subscribers or their designated representatives are the ultimate decision makers on how a DNS record for a phone number is to be provisioned.

How will the rights of telephone number subscribers be protected?

A simple answer is by respecting existing regulatory and business rules regarding number administration, slamming, non-reliance and so forth. Only by replicating or re-implementing ENUM analogs to the existing "rules of the road" will we avoid a wide range of serious administrative, operational and political conflicts.

How are you going to prevent "slamming" or "hijacking"?

Slamming, or the involuntary transfer of service providers, must be avoided in any ENUM system. It is a serious problem in the PSTN, and we must be careful not to expect more from Internet services than we are able to guarantee elsewhere.
Note that anti-slamming fundamentally requires a neutral third party solution. The U.S. industry is grappling with this issue with regard to long distance right now. It was solved on number portability from the outset.
Authenticated subscriber access is not a total solution, because if subscribers disconnect their telephony service, they lose rights to the phone number. Consequently, some combination of originator authentication as well as telephone number rights validation (using new and existing validation sources) can be used to solve the problem, depending on the level of standard required.

Are there any examples of global namespace delegation that should be considered as models?

The closest technical equivalent is in-addr.arpa. That domain provides a reverse mapping from IP address to domain name. It is used as part of the Internet infrastructure operation to help authenticate an IP address and identify the operator associated with an IP address. It is not seen directly by users. The same is true for e164.arpa. It will be for operational infrastructure, rather than for direct access by end users. As with e164.arpa, in-addr.arpa, allocations are hierarchical according to the infrastructure administrative structure. For in-addr.arpa, the hierarchy uses the "CIDR" address allocation hierarchy. For e164.arpa, the hierarchy will be based on the ITU E.164 recommendation.

effect of e164.arpa deployment on the global DNS

What is the effect of e164.arpa deployment on the global DNS system?

This is going to require research, such as the effect of "wrong dials" on the root of e164. That is, caller specification of a wrong number can result in many additional queries to the e164.arpa root.
Additional work will be necessary in advising ENUM applications about such things as the level of data caching necessary in order to relieve stress; suppress escalating of poorly-formed queries; misdials; or cache misses on the root structure. For telephony applications, performance and load engineering is critical, as query volumes from small- to medium-sized cities alone can easily reach many thousands per second. Response times, as well as transaction loads, must be carefully considered. Conventional DNS caching is of significantly reduced value in ENUM due to the huge size of the name space and relatively even distribution of queries into the space over arbitrary time intervals. Unlike conventional DNS queries, call volumes are not highly concentrated into a popular small subset of the number space.

How will the e164.arpa domain be organized?

How will the e164.arpa domain be organized?

One convenient way of doing this would be to delegate according to the ITU assigned country codes. It is important to understand, however, that delegation of a country code in DNS can occur at any digit or zone domain in DNS terms.
For example, within the root e164.arpa, there would be:
  • An NS listing for .1.e164.arpa - representing the country code (1) of the North American Numbering Plan (US, Canada, and several Caribbean countries).
  • An NS listing for .4.4.e164.arpa - representing the country code (44) of the UK.
  • An NS listing for .6.4.e164.arpa - representing the country code (46) of Sweden.
  • An NS listing for .1.8.e164.arpa - representing the country code (81) of Japan, and
  • An NS listing for .8.5.3.e164.arpa - representing the country code (358) of Finland.
At the national TN/NS level, further NS delegation [DNAME, CNAME, PTR] can occur with enterprises, TN/NS application service providers, carriers, and even individuals who have DNS servers in their homes.

Why use DNS?

The beauty of DNS is that it exists already and is a global, efficient, open and scalable system. Using DNS for ENUM also provides a low-cost solution.

How does ENUM work?

How does ENUM work?

A telephone number, once entered, is translated into an Internet address. The following steps explain the process:
  • The number entered is translated into a full qualified E.164 number. (To quote an example from earlier, the number 9777 1234 in Melbourne, Australia would be +61 3 9777 1234.)
  • The number is reduced to digits only. (e.g., 61397771234)
  • The digits are re-ordered back to front. (e.g., 43217779316)
  • Dots are placed between each digit. (e.g.,
  • The domain e164.arpa is added to the end. (e.g.,
  • A DNS query is then issued on this domain.
Once the authoritative name server is found, ENUM retrieves relevant NAPTR Resource records and will perform according to the user's registered services for that number.

What is .arpa, and why is it the top level domain for ENUM?
Why not create a new top-level domain specifically for ENUM?

E164.arpa has been designated as the temporary DNS domain for use with ENUM. This designation may change as a result of ongoing discussions between the ITU, the IETF and other international organizations involved with ENUM. (In the event that the international community chooses a different ENUM domain, the structures discussed here and in IETF RFC 2916 will apply to that new designated domain.)
The .arpa domain has been designated for Internet infrastructure purposes. It is managed by the Internet Assigned Numbers Authority (IANA) in cooperation with the Internet technical community under the guidance of the Internet Architecture Board (IAB). A new top-level domain (e.g., .e164) was not created because ENUM is an infrastructure application appropriate for designation within the previously established .arpa domain. ENUM is considered appropriate as an infrastructure application because it provides a set of DNS-based resource directories, referenced by phone number, for use by various ENUM-enabled application clients (such as telephones, SIP servers and voice messaging systems).

Why is the number reversed?

DNS, when reading a domain name, will first search for the top level domain (.arpa), then the second level domain (being e.164). It then searches for the country code, area code and phone number.

Why are there dots between the numbers?

Each dot separates the number into administrative domains, or zones. This allows for delegation of authority at various points along the name, and eliminates the requirement for clients to know individual delegation schemes to know where to put the dots.

Will ENUM telephone routing confuse the PSTN routing system?

ENUM will not affect PSTN application level functions such as call routing and signaling. ENUM facilitates the discovery of resources associated with a telephone number, and does not impact how applications operate.
It is important to note that ENUM will not change the existing right-to-use rules and principles for telephone numbers. ENUM is intended to facilitate applications using telephone numbers as subscriber names, not to change how telephone numbers are administered.

What are the main benefits of ENUM for subscribers?

What are the main benefits of ENUM for subscribers?

The primary benefit of ENUM is allowing Internet users the ability to communicate with other people via a range of services when the caller knows only the telephone number of that person or only has access to a telephone keypad. Internet services and resources can be accessed by end users with IP telephones, ordinary telephones connected to the Internet and other numeric-input, Internet-connected devices. ENUM allows end users to choose how they wish to be contacted, giving them greater control.

Who could use Public ENUM?

Aside from accredited Registrars, ENUM can be used by corporations, individuals, government agencies, military organizations and hosts of other non-individual users.

How is ENUM going to work for the average end user?
What will it look like?

The mechanics of using ENUM when a call is made from one person to another are virtually invisible. For instance, if one person called another and both parties were using IP phones, the call would appear as if it were being made over the regular PSTN. The calling party would designate the services its number would use, and the called party would also have specified which ENUM service they wish to have used to be contacted.

What kinds of applications could use ENUM?

The initial applications to which ENUM will be applied are Voice over IP (VoIP) and Voice Profile for Internet email (VPIM). The goal of the VoIP industry is to be able to make simple and high-quality voice calls using the internet, while the VPIM industry wants to develop voice mail systems that can exchange messages over IP networks. ENUM’s applications will most certainly reach beyond VoIP and VPIM to such services as Internet fax and Instant Messaging.

What does it mean to “provision” ENUM?

What does it mean to “provision” ENUM?

At the heart of the ENUM service is an Internet domain name registry.
Domain name registries have two core responsibilities: maintaining the authoritative database of domain name registrations, and publishing a "zone file" on the Internet that can be queried by any Internet user. The authoritative database has data about each domain name that is registered (who registered the name, contact information, the expiration date of the registration, etc.) The zone file contains information that will point the Internet user to the desired application, such as email or Web sites.
(For example, Neustar is the domain name registry for the .BIZ Internet domain. The Acme Company registers the domain name www.acme.biz with the .BIZ registry. As the registrant, Acme registers the name with the registry and is responsible for all adds, modifies and deletes related to this name. Further, when an Internet user types the address www.acme.biz into his or her Web browser, the browser will query the .BIZ zone file for a pointer to www.acme.biz, then it will query the Acme server to find the Acme Company Web page.)
In summary: To provision ENUM means to provision the administrative and pointer information for the ENUM domain name with the ENUM registry. For example, if someone's phone number was +1 202-555-1212, he or she would provision the domain name with the registry that is responsible for the domain That would make him or her the registrant for that domain name.

What is Private Infrastructure ENUM?

Private ENUM uses the concepts of creating a domain name from a telephone number (TN) and resolving it to a URI. It does not, however, use the domain e164.arpa (or any other domain with regulatory oversight). Private ENUM is used when there is a closed user group that wants to use ENUM to exchange IP traffic.
An example of such a user group is a group of communications carriers that wants to exchange VoIP traffic. The user group that creates and uses the domain will also create the policies for the domain. There will be no regulatory oversight of this domain, because it is not intended for public use.

What is Public Infrastructure ENUM?

National number administrators typically assign TNs to communications carriers, not to end users; carriers then assign the TNs to end users. Carriers have specific needs when it comes to provisioning routing information related to their customers’ TNs. Carriers map TNs to internal network addresses to enable call routing and features; these addresses are not publicly available. The carrier’s network and its associated addresses are highly secure, and access is strictly maintained and limited to other service providers. These characteristics have come to define the recent discussions of carrier ENUM.

What is Public User ENUM?

With Public User ENUM (also called "End User ENUM"), the end user can provision his or her records in the ENUM registry in the public domain e164.arpa. In other words, the end user is the registrant for the ENUM domain name within the domain e164.arpa; the implication is that the end user has "opted in" to the ENUM service.
"End User opt in" is generally considered a good policy if end user data is publicly available on the Internet. (The DNS is publicly available to any Internet user; it uses a domain available on the public Internet, and various regulatory, industry and standards bodies moderate the domain’s policies and practices.)

What is the ITU?

The International Telecommunications Union (ITU) is a global organization where governments and the private sector come together to coordinate global telecommunications networks and services. The ITU administers E.164 (described above) as the international country-level telephone numbering plan, the structure of which is described in the ITU document "E.164."
Country codes are assigned by the ITU; however, designated telecommunications regulatory bodies within each country administer their own telephone numbers. (An example of a fully qualified E.164 number includes that telephone number's country code, area code and the phone number. For example, the phone number 9777 1234 in Melbourne, Australia would be +61 3 9777 1234.)

Android 4.1, Jelly Bean

Android 4.1, Jelly Bean: The world's most popular platform gets even better

Android 4.1, Jelly Bean, is the fastest and smoothest version of Android yet. Jelly Bean improves on the simplicity and beauty of Android 4.0, and introduces a new Google search experience on Android.

The new Google experience on Android 4.1

The new Google experience on Android

The best of Google just got better on Android. The search experience in Jelly Bean has a new look-and-feel, and has faster and more natural Voice Search. And Google Now brings you just the right information, before you even ask.

A new look for Search.

Android has search at its core. With Jelly Bean, a redesigned experience uses the power of the Knowledge Graph to show you search results in a richer way. It's easier to quickly get answers and explore and browse search results.

Voice Search.

Sometimes you'd rather just speak your search query. Or just ask a question. Android lets you search the web with your voice, and it's convenient for getting quick answers on the fly. It speaks back to you and is powered by the Knowledge Graph, bringing you a precise answer if it knows it, and precisely ranked search results, so you can always find out more.

Google Now brings you just the right information at just the right time.

Google Now tells you today's weather before you start your day, how much traffic to expect before you leave for work, when the next train will arrive as you’re standing on the platform, or your favorite team's score while they’re playing.
And the best part? All of this happens automatically. Cards appear throughout the day at the moment you need them. Learn more

Android Beam

Android Beam.

With Android Beam on Jelly Bean you can now easily share your photos and videos with just a simple tap, in addition to sharing contacts, web pages, YouTube videos, directions, and apps. Just touch two NFC-enabled Android devices back-to-back, then tap to beam whatever's on the screen to your friend. Instantly pair your Android phone or tablet to Bluetooth® devices like headsets or speakers that support the Simple Secure Pairing standard by just tapping them together – no more syncing or searching required.

Android 4.1: Better Accessibility


With Jelly Bean, blind users can use 'Gesture Mode' to reliably navigate the UI using touch and swipe gestures in combination with speech output. Jelly Bean also adds support for accessibility plugins to enable external Braille input and output devices via USB and Bluetooth.

Android 4.1: A smarter keyboard

A smarter keyboard.

Android's dictionaries are now more accurate, more relevant. The language model in Jelly Bean adapts over time, and the keyboard even guesses what the next word will be before you've started typing it. With improved text-to-speech capabilities, voice typing on Android is even better; it works even when you don't have a data connection, so you can type with your voice everywhere you go.

Android 4.1: Seamlessly take and share photos.

Seamlessly take and share photos.

Android 4.0, Ice Cream Sandwich, made snapping photos super fast; Jelly Bean brings that same speed to the next step: viewing. Just swipe over from camera to filmstrip view to instantly view the photos you just took, and quickly swipe away the ones you don’t like. Now sharing — and bragging — are a breeze.

Android 4.1, Jelly Bean: Android gets even better

Fast & smooth

We put Android under a microscope, making everything feel fast, fluid, and smooth. With buttery graphics and silky transitions, moving between home screens and switching between apps is effortless, like turning pages in a book.
More reactive and uniform touch responses mean you can almost feel the pixels beneath as your finger moves across the screen. Jelly Bean makes your Android device even more responsive by boosting your device's CPU instantly when you touch the screen, and turns it down when you don't need it to improve battery life.

Simple, beautiful and beyond smart

Expandable, actionable notifications.

Android has always put you in control when it comes to staying notified and connected. Now you can take action directly from the notifications shade. Late for a meeting? Email everyone to let them know. Missed a call? Call them back in an instant. And because they’re expandable, you can get an even deeper look into the things that matter most, like multiple emails or photos on Google+.

Widgets work like magic.

With Jelly Bean it's now even easier to personalize your home screen. As you place widgets on the screen, everything else automatically moves to make room. When they're too big, widgets resize on their own. Interacting with your favorite apps and customizing your home screen has never been easier.

Free SIP Providers

Having a free SIP account is a great way of making free calls on the Internet. You only need to choose a SIP provider that gives you a SIP account for free. There are many of these. Here is my list:

1. GetOnSip

GetOnSip is a straight service offered by Junction Networks, the company behind OnSip.com, offering free SIP accounts in a very simple and easy registration. The registration fields stand right in your face as soon as you load the home page. You are not restricted on the amount of SIP accounts, and you get free SIP voice and video calling service and XMPP messaging. GetOnSIP also offers paid business SIP accounts.

2. IPTel

IPTel provides IP telecommunications services and hosts several projects like the SIP Express Router, SIP Express Media Server and the SIP Express Router Web. They also provide on their site a wealth of information on Sip communication. The free SIP account they offer is of good quality and is also straightforward to register.

3. Ekiga SIP Service

Ekiga is a VoIP service that offers a softphone and free VoIP service, and it is known for its Linux VoIP app. Ekiga also offers free SIP addresses that you can use with any SIP-supporting softphone app.

4. Sip2Sip

Sip2Sip is a rather straightforward SIP service, offer by ag-projects.com. It is a free SIP service based on fair use policy and no support. Registration and account management are easy. AG Projects has lots of products and offers this free SIP service for users to test the features present in its products.

5. AntiSIP

AntiSIP offers a set of SIP-based services among which is the free SIP account, to which you can even receive landline calls, for example through the free US numbers provided by IPKCALL.

6. VoIPUser

VoIPUser SIP account is a community effort of VoIP enthusiasts. Do not expect much support from them. The registration process, while being straightforward, is rather restrictive. But if you believe in community efforts, check them out.

Installing and Configuring 3CXPhone for iPhone

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Installing and Configuring 3CXPhone for iPhone

By November 11, 2010 | 27 Comments


  1. Click here to be taken to the 3CXPhone for iPhone within the Appstore, or start up the Appstore manually
  2. Search for 3CXphone or just type in 3CX. This search will bring up the 3CXPhone for iPhone
  3. Select install
  4. The phone will be installed automatically.

Configuring 3CXPhone for iPhone

After you have installed 3CXPhone for iPhone, you must configure it to use a VoIP provider or a VoIP PBX (SIP server) in order to make and receive calls. Note that 3CX only provides the software, we do not provide any calling services. After you have obtained an account at a VOIP provider or your Extension details from a VoIP PBX, you must proceed to configure a connection to that account in 3CXPhone as follows:
  1. Start 3CXPhone by tapping the 3CXPhone icon. The first time you start 3CXPhone, it will automatically prompt you to create a new connection profile or to use provisioning. To create a profile manually click on “Create Profile” and go to (3)
  2. If you are connecting to 3CX Phone System and you are located on the same LAN (in the office) you can have 3CX Phone for iPhone automatically configured using Auto Provisioning. More information available here (3CX Phone System V10 and higher is required).
  3. Configure Name and authentication information
    • Name: Configure a name to identify this account.
    • Display Name: Enter your first and last name here. This is the name that other phones will see on their display.
    • User: Enter the extension number or user ID (sometimes referred to as Extension number, SIP ID, Phone number, Account)
    • ID: Enter your Authentication ID (sometimes referred to as Auth ID, Authentication Name, User name, Authentication user name).
    • Password: Specify your password in the ‘Password’ field.
      NOTE: The terminology used may vary between VoIP Providers and PBXs.
  4. Configure Use As Toggle
    There are three states 1) In Office 2) Out of office Tunnel 3) SIP Account

    a) If you are connected to a 3CX Phone System and you are in the same network, select In Office
    b) If you want to connect to 3CX Phone System and you are out of the office select Out of office Tunnel. For more information see Configuring 3CXPhone for iPhone with 3CX VoIP Tunnel
    c) If you want to connect to a VoIP Provider select SIP Account
  5. In the Local PBX IP field enter the IP Address of your PBX or SIP Server example If you are connecting 3CXPhone to a VoIP Provider you can skip this step.
  6. In the “External PBX IP” field, specify the domain name or the public IP of the SIP server (also referred to as registrar server or SIP registrar) of your VoIP Provider or VoIP PBX. For example, sip.myvoipprovider.net.
  7. If your VoIP provider has given you a STUN server name, specify it in the “STUN Server” field here. If not, you can leave the STUN server entry as stun3.3cx.com
  8. If your provider needs a proxy server enter the address in the Proxy field
  9. Click DONE to create the connection profile. This new profile will be displayed in the “Profiles” page. Once you have logged in successfully, your user name will be shown at the top on the right hand side

Fixing Android Tablet Not Booting A Personal Experience With Chinese Tablet

I am writing this tutorial as a personal experience with my chinese-made tablet pc. In general it is a 7" MID ePad Android Tablet powered by VIA 8650 800MHz CPU and Android 2.2 Froyo operating system. I believe the built in operating system is supplied by WonderMedia Technologies (WMT) as there are many WonderMedia Technologies applications pre-installed in it and some of them come with chinese user interface which i cannot understand and decide to uninstall later.
I decided to buy this chinese tablet because it was very affordable compared to other currently famous tablet pc such as Apple iPad, Samsung Galaxy Tab and Dell Streak, which price may be 4-5 times higher. And this was also my first time making purchase from overseas which i supposed to be quite risky. But considering the price was cheap (under $100) and if any bad things happened (cheating seller, broken in shipment, undelivered and etc) i just lost some bucks so i finally decided that it was worth to try and i will take the risk.
First Encounter
The parcel was delivered in about 10 days by UPS ExpressSaver which I had chosen when making the purchase. When the courier arrived at my home, I was asked to pay the import duty fee charged by customs and it cost me about $28. But later, after analyzing the import duty payment form, I found out that the handling fee charged by UPS itself cost more than the customs..!! Though I was disappointed, still I was very happy to receive this long awaited tablet pc in good condition. It was well packed in a carton box, wrapped with bubble form and put in a plastic UPS polybag.
I cant wait to open the package and turn on my tablet for the first time. It was a very happy moment like the moment when Archimedes cried out "Eureka!!". The android 2.2 froyo operating system started with an animation of android robot dancing and flying on the center of the screen then booted successfully to the home screen (in Windows we call it Desktop). The next few weeks after I received my tablet pc, I found myself playing games, typing emails, browsing sites, using facebook, playing multimedia contents and installing lots of applications with this chinese tablet pc. To add new android applications to this tablet, you cannot use the regular android app market such as found in Samsung GalaxyTab because it needs a registered google account with the android OS. I think that this chinese tablet is not supplied with legal, registered Android OS. The only way to install android applications to this chinese tablet pc is by downloading .apk extension files from internet, put them in your SD card and install them with the built-in file browser pointed to that SD card. Apk extension stands for Android Package and it is the installer file you need to add application to your chinese tablet pc, idem to msi extension in Windows OS. Running the apk files will install the application. Some places to download APK (android installer) files are:
Nightmare: Tablet Not Booting  

There was a night when my gf played angry birds in my chinese tablet pc and the next morning when I turned it on, it quitted booting and the screen froze with the android robot stood still in the center of the screen while on the left bottom corner of the screen showed: Android TM OS 2.2
Kernel 2.6.32
Build Number V1.3.1

Android Tablet Freeze on Booting Process

Trial I: Unsuccessful
I had tried a lot of ways to make it boot again including hard power off (pressing the power button for more than 5 seconds), reset the device (pushing a small pin to the small hole at the side of the screen, there are 2 holes: one for microphone and the other for reset button, you should try which one), leave it turned on until the battery drained off (just like pulling a bios battery off a motherboard to reset), try to press combinations of various keys I can find on the tablet pc. They all failed to fix this chinese tablet pc and made me start thinking that all chinese products are lack of quality. I searched google and read a lot of forum posts but find no solutions to my problem. I also tried to upgrade the firmware by copying new firmware to \script\ folder on the root of my SD card and put the SD card in its slot before turning on the tablet pc. It failed too because the screen did not react nor show anything, just freezing on the first boot screen with the android robot posing still in the center of the screen. Since then I always hate that robot. It was a frustrated experience and I started to read from forum posts that it could be a problem with the dead NAND drive in the tablet or the overheating/overcharging battery which may cease working.
Trial II: Failed Too
I tried to contact the seller and asked for solution to my problem, hoping that I could fix it myself without sending it back the country where it came from because if I did that, I had to bear the two ways shipping costs as one of the warranty terms and frankly they were not cheap and the customs may charge me another import duty fee when the repaired package arrived at my doorway. Overall the shipping costs may cost me the same price as a new tablet pc.
The seller is good and very cooperative, one of their sales team, a woman, told me to try to upgrade the firmware and provided me with 3 links of the firmware archive files. The 3 files were part of a RAR file which were splitted. The links were hosted in a chinese file sharing site :
EP07-V2.2.part1.rar -
EP07-V2.2.part2.rar -
EP07-V2.2.part3.rar -
As instructed, I downloaded all the files and used WinRAR to extract all the firmware files to \script\ folder in my micro SD card (some people call it TransFlash). So if my micro SD was loaded to drive F:\ then all the firmware files were stored in F:\script folder. Then I removed my micro SD card from the card reader and put it in the tablet micro SD slot while the tablet is off. With the power adapter kept charging to ensure that no power failure during firmware upgrade process, I turned the tablet on, hoping to get a message or warning of the firmware upgrade process like the one described in one forum about the firmware upgrade process. The first frame I saw on screen was the standing still android robot again, but after a minute, the screen did not shift to other frame about warning nor message. Curious, I waited longer (20 minutes - 30 minutes) and there was still nothing on the screen. For me, it meant there was no hope for my tablet, it had totally died. I was very upset and threw it to a corner of my house. "Rubbish" I mumbled. Due to my busy activities at that time, I had almost forgotten it and did not even try to contact the seller again to tell her about the failure.
Two weeks later, I had been through the busy times and that time I tried to make another contact to the seller again. Another sales team (not the one who handled my case before) replied my message. She asked me about the type of tablet I purchased and offered to upgrade the firmware again, but this time with a different firmware, not the previous two files. She said that was a new firmware suitable for my chinese tablet. I did not believe her anymore and said that I had tried upgrading the firmware given by her coworkers before and it failed. I also told her that there might be a problem with the internal NAND drive or overheating battery or processor as all tablets are sealed with no air circulation for its internal electronic components.
She told me not to worry (but I actually DID..!!), and asked me if I wanted to try her suggestion first or sent the tablet back to them immediately. Considering the high two-way shipping costs plus another customs import fee, I finally made myself try her suggestion with the latest firmware. I downloaded it from a link she provided:
New Firmware:
Official FW M009s WM8650 Android2.2 7" xx.11.06(Mediafire) -
Alternative Link:
Official FW M009s WM8650 Android2.2 7" xx.11.06(Rapidshare) -
She also provided me with old firmware in case i need it :
7inch 8650 latest firmware part 1 2.zip
Root patch part 2 2.zip
I never used this old firmware, i used the new one.
After I successfully downloaded the new firmware (Official FW M009s WM8650 Android2.2 7" xx.11.06) and viewed the archive with WinRAR, I saw a different file structure compared to previous firmware:
The first and second lines were folders, the last one was a file. I tried to view the content of "wmt_scriptcmd" in Notepad and found out that the content had a lot of line pointed to FirmwareInstall\*.* files. I made a summary that this "wmt_scriptcmd" was probably like "autorun.inf" file in Windows OS which directed execution to certain files in FirmwareInstall folder. There was no "script" folder anywhere in the archive, contrary to what I knew about the firmware upgrade process which involving "script" folder.
Next, I copied all the contents of M009S_06.11 folder to the root folder of my micro SD so they became:
Then I removed the micro SD from card reader and inserted it in the tablet micro SD slot. For safety and to make sure there was no battery failure during the firmware upgrade process, I plugged in the power adapter. After that I turned on the device and as usual, the android robot showed up. I did not expect this, but maybe the robot knew I hated it, this time it was just awhile and the screen shifted to next message showing the upgrade process. "Eureka..!!" I finally can upgrade the firmware, but still I did not know what would happen next, would the firmware upgrade process fail in the middle? and if the upgrade was successful, would it fix my tablet ? Here is the screenshot of the firmware upgrade process:
Android Firmware Upgrading Process
Here is the documented android firmware upgrade process step by step, as shown on the screen:
Upgrading W-Load to SF.
Upgrading U-Boot to SF.
Upgrading U-Boot env.cfg. to SF.
Upgrading Ramdisk to NAND.
Upgrading Kernel to NAND.
Upgrading FileSystem partition(Nand).
   copy rootfs patch&restore files...
Upgrading U-Boot LOGO to NAND
Upgrading Kernel LOGO to NAND
copy data to Android-Data partition(Nand)..
copy pre-files to LocalDisk...

On top of the screen there is a message "WM8650 Android2.2 FW Upgrading ..." while on the bottom of the screen there is a progress bar and a message "DON'T REMOVE INSTALLATION MEDIA" which changes to "PLEASE REMOVE INSTALLATION MEDIA" after successful upgrade.
If you are interested to see the video of the firmware upgrade process, i have successfully documented it here:

The upgrade process was quite fast, it just took less than 4 minutes to finish. After the process finished, the screen displayed a message to remove the micro SD and restart the tablet. After restarting the tablet, here came again the android robot, but it was just standing for a while and then proceeded to dance on the center of the screen. I was really happy to find out that this android robot had eventually come to life again after 2 months of inactivity. What an experience.. I believe that there are still a lot of people out there who may have the same problem like mine, frustrated with unbootable chinese tablet device. I personally hope this article will save your time, effort and money to fix your unbootable chinese tablet by trying what I did with my tablet.
Summary: Never Give Up
The key is to use the correct and suitable firmware for your tablet, though all the firmwares are built with Android OS but they may vary by brands and models. One firmware may be suitable for a tablet but may not be suitable for others. To look for the correct firmware, I recommend you to refer to your seller or product manufacturer, either by using their online website or by contacting their customer service personally.

Footnote: Resolving Malfunctioning Touch Screen Issue After Successful Flashing and Booting

In certain cases when you have successfully flashed the new firmware and made the android tablet boot normally again, but find out that the touch screen is not functioning and not reacting to any of user touch input, it is caused by a different touchscreen decoder chip you have in your tablet. To make it works for your tablet, you need to edit some of the firmware files. You can try this solution pointed out by one of our commenters ShibuVas:
For WM8650 models follow this to resolve touchscreen issue. There are many different versions of 8650 tablets using different touchscreen decoder chips - vt1603 or uor615x or uor615 or vt1609 or it7260 or others. The following example shows you how to modify the touchscreen decoder chip model in firmware files from vt1603 model to uor615x model. Make the following changes to files in the SD card prior to flashing:
  1. file FirmwareInstall/packages/fs_patch/root_patch/system/default.prop :
    find the line ro.wmt.touchpannel=0 and change it to ro.wmt.touchpannel=1
  2. file FirmwareInstall/env/uboot_env_head, for the 2 lines :
    setenv touchcodec vt1603 and setenv wmt.io.touch 1:vt1603:2:800:480
    change them to,
    setenv touchcodec uor615x and setenv wmt.io.touch 1:uor615x:0:800:480
Then flash the firmware as usual. If you have an 8" tab change all the '480's to '600's d( 800x600 resolution for a 8" tablet) - not applicable to your 7" case.
Currently documented touchscreen decoder chips are:
»   vt1603
»   vt1609
»   uor615x
»   uor615
»   it7260

Documented audio decoder chips are:
»   vt1603
»   wm8900

Please note if the touchscreen still does not work for you, i believe that you may have other touchscreen decoder chip type which we have not discussed here. I suggest you to ask your seller for the tablet specification especially the touchscreen decoder chip so you can alter the firmware files based on it.
Good luck, adios..

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