Happy New Year 2011

How to Use a Linux PC to Send and Receive SMS Messages

How to Use a Linux PC to Send and Receive SMS Messages

Cisco makes VoIP call from space

Cisco has made the first ever VoIP call from space through its latest satellite technology.

By installing an internet protocol (IP) router onboard an already orbiting spacecraft, the networking specialist was able to make the connection without the use of any ground-based infrastructure.


Article from Telappliant

VoIP Trunk Registration

The SIP trunk which we are configuring will first send the credentials to the server, from the server we will get response 401 message then from the useragent we have to send the credentials with auth name then only the VoIP trunk will get registered.

Embedded software developer|Embedded system course and training HYD

Embedded software developer|Embedded system course and training HYD

Java 7 and 8 Begin to Take Shape

Java 7 and 8 Begin to Take Shape: What's In, What's Out?

The next two major versions of Java are beginning to take shape. Oracle has submitted the release content for both Java SE 7 and 8 to the Java Community Process (JCP) for approval before the end of the month.

JSR (Java Specification Request) 336 details the components and features that will become JDK 7, while JSR 337 details JDK 8. The final Java 7 release is being targeted for mid-2011, while Java 8’s release is set for 2012.

An article from www.developer.com

Sangoma Introduces the Highest Density Voice Transcoding Card Available on the Market Today

The industry's highest density transcoding card — designed for optimum voice quality for high port count systems, offering from 400 to 2,000 transcoding sessions in one compact PCIe form factor! For more detail and data sheet Click Here

Iron Speed Designer - Build Database & Reporting Web Applications Without Coding! - Overview

Iron Speed Designer - Build Database & Reporting Web Applications Without Coding! - Overview

What is the difference between a domain and web hosting?

What is the difference between a domain and web hosting?

Free Tool to Test Your Internet Connection For VoIP Packet Loss

Free Tool to Test Your Internet Connection For VoIP Packet Loss

Asterisk 1.8 Secures Open Source VoIP

"Finally, there is a new long-term support (LTS) version of the open source Asterisk VoIP IP-PBX system. Asterisk 1.8 is being released this week, marking the first new LTS for Asterisk since the 1.4 release in 2006 .

"The new Asterisk 1.8 release is intended to be supported for at least the next four years, as part of a new support model the project first discussed earlier this year. Asterisk 1.8 packs in a long list of new features, including reverse call display and integrated Google Voice support.

""This is huge news for people that are building commercial systems or their own phone system out of Asterisk, as they'll get four years of support from Digium," Steve Sokol product manager at Digium told InternetNews.com. "So if people are looking to jump in and start developing a solution based on Asterisk, this is the right time."

Article from LinuxToday

Voip – Advantages And Disadvantages

The new technology of VoIP has caught on big time and many business have completely gone over to the other side. With complex call routing, and intelligence built in, Voiceover IP has definitely showed itself to be the future of telephony. However, there are still many wrinkles to be ironed out and in this article we take a look at both the advantages and disadvantages of VoIP networks.

Advantages of VoIP

There are three big advantages that VoIP enjoys over traditional PSTN systems. Cost, flexibility and scalability. The first one, cost, is the most obvious and is the biggest pull for individual customers these days. A VoIP call can be made at a fraction of the cost of a regular PSTN connection. When we look at overseas calls, it gets even better. The rates which VoIP allows providers to charge are almost negligible. A huge cost saving for those who need to talk to business partners or loved ones overseas for long periods of time.

The second advantage – flexibility – refers to the ability of businesses and providers to integrate a VoIP system into any number of applications to any extent depending on the skill of the programmer. Therefore it can be tied into sales and CRM data as well accounting systems. This can vary from place to place depending on the needs. Additionally, complicated rules can easily be created regarding call routing, business hours, conferencing and any other use that VoIP is put to.

Finally, VoIP inherits the scalability of the Internet. Adding a new user is in many cases as simple as plugging in the phone and creating an account. Just like an email box. And the costs don’t go up the way they do with PSTN systems.

Disadvantages of VoIP

Many VoIP networks are still not interoperable. Meaning that Skype users for example have to utilize the PSTN network in order to communicate with another VoIP system such as SIP. This increases the cost of calls and forgoes benefits such as HD voice which rely on high end codecs sent entirely over IP.

VoIP call quality has been improving dramatically and in many cases is superior to PSTN, but there is still the occasional glitch caused by the fact that the Internet is basically stateless.

Finally, the standards used by VoIP are still being standardized. In a few years more, it should be pretty much settled what everyone is going to use. But until that time, we have to go with what most people work with – SIP – and hope that others catch up soon.

An article from Virtual Office 

OpenBTS with Asterisk

BTS - Base Transceiver Station also called as Radio Base Station (RBS) is a device that connects user end handheld device (mobile phones) with network (GSM, CDMA, etc...)


OpenBTS replaces the traditional GSM operator network switching subsystem infrastructure, from the Base Transceiver Station (BTS) upwards. Instead of forwarding call traffic through to an operator's mobile switching centre (MSC) the calls are terminated on the same box by forwarding the data onto the Asterisk PBX via SIP and Voice-over-IP (VoIP). More Details

Asterisk vs. FreeSWITCH

Asterisk vs. FreeSWITCH

OpenSIPS and Asterisk Compared

OpenSIPS and Asterisk Compared: "Back to Back User Agent"

How to Secure a VoIP | eHow.com

How to Secure a VoIP | eHow.com

VOIP Hacker Gets 10 Years in Prison for Scam - Security from eWeek

VOIP Hacker Gets 10 Years in Prison for Scam - Security from eWeek

Smartphones Are Turning Us Into Idiots | ZDNet

Smartphones Are Turning Us Into Idiots | ZDNet: "that could act as a wifi access"

Cacti: The Complete RRDTool-based Graphing Solution

Cacti Can also be used by integrating with Asterisk AMI (Asterisk Manager Interface)... More details are provided in the linkCacti: The Complete RRDTool-based Graphing Solution

Cumbersome Asterisk

Article from HERE...
Asterisk issues and bugs I have found in the software
I ran into some problems when I tried to deploy Asterisk on a VPS first (Amazon EC2). When I tried to do conferencing, I needed the MeetMe application that provides conferences, and MeetMe depends on Zaptel (which is another piece of software that provides the timing that MeetMe needs). I tried to install Zaptel, but Zaptel refused to work on XEN (the virtualization software that our VPS uses). So I didn't have any other choice but to go with a dedicated server for our VoIP needs. FreeSWITCH works great on a VPS with all the conferencing features, and everything out of the box - no zaptel, ztdummy or anything that could interfere is needed. More Details

Samsung takes on the Apple iPad with the 7 inch Galaxy Tab device | ZDNet

Samsung takes on the Apple iPad with the 7 inch Galaxy Tab device | ZDNet: "- Sent using Google Toolbar"

Earn Money

A Indian company earn money by clicking ads in this website.

Referal Link http://www.newbuxindia.com/?ref=saigop 

The World of VoIP, Asterisk and Linux

The World of VoIP, Asterisk and Linux: "- Sent using Google Toolbar"

How Consultants Help SMBs Choose VoIP Systems - VoIP News

How Consultants Help SMBs Choose VoIP Systems - VoIP News: "- Sent using Google Toolbar"

10 Tips for the First-Time Business Owner | Insight | Mumbai | India

10 Tips for the First-Time Business Owner | Insight | Mumbai | India: "- Sent using Google Toolbar"

SIP - One Way Audio

In SIP if you work with Nat, Firewall, some time you will be facing one way audio, for which you need to open some ports in firewall. TCP port 5060 and UDP ports 5000 to 31000. Apart from this you can also enable stun server if you face still one way audio. More information click here...

Seven Steps to Better SIP Security with Asterisk

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, Asterisk-based systems included. 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. You can take steps, NOW, to eliminate many of these problems. I think the community is interested in coming up with an integrated Asterisk-based solution that is much wider in scope for dynamic protection (community-shared blacklists is the current thinking) but that doesn’t mean you should wait for some new tool to defend your systems. You can IMMEDIATELY take fairly common-sense measures to protect your Asterisk server from the bulk of the scans and attacks that are on the increase. The methods and tools for protection already exists – just apply them, and you’ll be able to sleep more soundly at night.

Seven Easy Steps to Better SIP Security on Asterisk:

1. Don’t accept SIP authentication requests from all IP addresses. Use the “permit=” and “deny=” lines in sip.conf to only allow a reasonable subset of IP addresess to reach each listed extension/user in your sip.conf file. Even if you accept inbound calls from “anywhere” (via [default]) don’t let those users reach authenticated elements!
2. Set “alwaysauthreject=yes” in your sip.conf file. This option has been around for a while (since 1.2?) but the default is “no”, which allows extension information leakage. Setting this to “yes” will reject bad authentication requests on valid usernames with the same rejection information as with invalid usernames, denying remote attackers the ability to detect existing extensions with brute-force guessing attacks.
3. Use STRONG passwords for SIP entities. This is probably the most important step you can take. Don’t just concatenate two words together and suffix it with “1″ – if you’ve seen how sophisticated the tools are that guess passwords, you’d understand that trivial obfuscation like that is a minor hinderance to a modern CPU. Use symbols, numbers, and a mix of upper and lowercase letters at least 12 digits long.
4. Block your AMI manager ports. Use “permit=” and “deny=” lines in manager.conf to reduce inbound connections to known hosts only. Use strong passwords here, again at least 12 characters with a complex mix of symbols, numbers, and letters.
5. Allow only one or two calls at a time per SIP entity, where possible. At the worst, limiting your exposure to toll fraud is a wise thing to do. This also limits your exposure when legitimate password holders on your system lose control of their passphrase – writing it on the bottom of the SIP phone, for instance, which I’ve seen.
6. Make your SIP usernames different than your extensions. While it is convenient to have extension “1234″ map to SIP entry “1234″ which is also SIP user “1234″, this is an easy target for attackers to guess SIP authentication names. Use the MAC address of the device, or some sort of combination of a common phrase + extension MD5 hash (example: from a shell prompt, try “md5 -s ThePassword5000″)
7. Ensure your [default] context is secure. Don’t allow unauthenticated callers to reach any contexts that allow toll calls. Permit only a limited number of active calls through your default context (use the “GROUP” function as a counter.) Prohibit unauthenticated calls entirely (if you don’t want them) by setting “allowguest=no” in the [general] part of sip.conf.

Opening Up with Google

Although Google is the most popular Web-based search engine used by everybody, not many know that it’s one of the biggest supporters of open source. For more on Google’s association with open source, here’s the story.

In January 1996, Larry Page and Sergey Brin, both students at Stanford University, began working on a search engine, which came to be called Google and took the world by storm. Initially, Page and Brin extensively used open source software such as Linux and GCC, internally. One of the reasons for using open source was, perhaps, the cost savings. However, the power that open source offered them—the ability to study and customise the source code to their preference—played a bigger role.

The creators of Google used free software wherever possible, even for corporate processes like accounting, record-keeping, etc. The use of free software for Google’s systems continues till date, with nearly all of its servers running on Linux. Page and Brin also used free software libraries such as OpenSSL, zlib, PCRE and MySQL. Since Google depended extensively on free software, its creators felt obliged to contribute to the various open source projects that had led to Google’s success.

If the software improves, everyone benefits, including Google. So, why keep the in-house fixes and improvements to the code a secret? Click Here for more....

A Quick List For VoIP Knowing | VoIP Websites

A Quick List For VoIP Knowing | VoIP Websites: "- Sent using Google Toolbar"

SIP Client for Blackberry

Callto.Net BlackBerry Vippie!

Difference between SS7 and ISDN PRI

Both are similar in many ways, but SS7 is more powerful and core to the network. PRI is a subset of SS7.

ISDN PRI link must directly be connected to switch. It has no capability to route the call intelligently. Using SS7 every node in the network can be accessed. Nodes need not be connected directly, SS7 has got built in intelligence to redirect packets to the correct destination.

ISDN PRI is divided into Transport (B) channels and signaling (D) channels. Voice/Data is transmitted through the B channels and the associated signaling information is transmitted through the D channel. T1 has 23B and 1D channels. E1 has 30B and 2D channels.

SS7 allows to access the telecom companies databases, like the subscriber data, and other configuration information. Now a days most of the call routing is based on database information about the customer and the destination number.

An example for this would be, the credit limit approaching message that we hear when our credit limit is about to expire. Whenever a call is placed, subsriber information is looked up from the database the appropriate message is played or routed to the destination.

SS7 is used to interconnect the switches. Its not a consumer end protocol. Modern day switches convert SS7 to PRI before giving it to a vendor or a consumer.

The difference between PRI and SS7 is the kind of signaling that occurs. SS7 is the signaling network for telcos. Like ISDN, inter-telco signaling is OOB. However, SS7 links carry only signaling. And signaling is not solely related to call passing. LNP (Line Number Portability) transactions occur on this kind of link. You may think of ISDN as an IGP and SS7 as an EGP. ISDN is for processing calls between the telco and the customer. SS7 is for processing signaling, generally between telcos.

Encrypt Your Mobile VoIP Calls

Whisper Systems, announced the availability of it public beta Mobile Security Suite, with two applications for Encrypting SMS and VoIP calls on Android devices.

The VoIP application uses a new method of establishing a call using SMS as a signaling protocol instead of the initial SIP signaling, to overcome the SIP constant connection requirements.

The encryption is done using the well-known ZRTP protocol to setup SRTP stream between the 2 devices.

Here is how it works:

• The caller uses the RedPhone software normally to call any RedPhone number, the software will contact the RedPhone infrastructure, which will send an encrypted SMS to the destination RedPhone device.
• The received SMS will activate the RedPhone VoIP client on the destination, and the call will then start the normal sequence.

The 2nd application is TextSecure, which uses OTR encryption protocol with ECC.

Pretty neat idea, but it is only available in US for now, and only on Android devices.

Asterisk - Building a T1/E1 Loopback Connector

In order to perform a loopback test of the E1/T1 port, a loopback plug (also known as loopback connector or loopback cable) is needed. An E1/T1 loopback connector can easily be made using a single RJ45 connector and two 4-in (10-cm) wires.

Materials Needed:

• Unused RJ-45 plug
• Two strands of Category 5 (Cat-5) wire, each 4-in (10-cm) long
• Crimp tool for RJ-45

Procedure:

1. Insert one end of wire 1 into pin 1 of the RJ45 connector. Note: To ensure a good connection, make sure that each wire goes all the way into the end of the plug. Refer to Figure 1 for an example of an RJ45 pin layout.
2. Insert one end of wire 2 into pin 2 of the RJ45 connector.
3. Insert the other end of wire 2 into pin 5 of the RJ45 connector.
4. Crimp the connector.

Now you have a small loopback cable: a single RJ45 connector with pin 1 connected to pin 4, and pin 2 connected to pin 5.

After you install the loopback cable, zttool (or dahdi_tool) should show the span status as "OK" (green LED) on that specific span. If the status of that specific span is not "OK", then the E1/T1 loopback cable was made incorrectly (try another loopback cable), or the port is defective.

Smartphones to be used as hotel room keys

The key card could become a thing of the past after a hotel chain announced it would allow guests to access their rooms using their smartphones.

The technology, to be trialled at two hotels next month, would mean that guests could choose to avoid the hassle of checking in at the front desk.

Instead, they would download an application to their mobile device that would enable them to open their door simply by holding their phone to a sensor.

Testing will take place for at least 60 days at the Holiday Inn Chicago O'Hare Rosemont and the Holiday Inn Express Houston Downtown Convention Center, starting next month.

Bryson Koehler, an executive at InterContinental Hotels group, told USA Today: "The holy grail item for us is to simplify the room-key hand-off moment at the hotel.

"We don't need to burden people with additional items; it just clutters up their lives. The beauty of the smart phone is that they've already got it."

He said the technology would be compatible with most smartphones, including the iPhone, BlackBerry and Android phone.

If the trial is successful, it could be extended across the company's hotel network with sensors fitted to about one in five of each hotel's rooms.

Android 2.2...Just Image...

New king of tech: Apple overtakes Microsoft

New king of tech: Apple overtakes Microsoft

Flying cars may soon be a reality - Yahoo! India News

Flying cars may soon be a reality - Yahoo! India News

Boy fixes cracked iPhone screen for $21.95 (photos) | ZDNet

Boy fixes cracked iPhone screen for $21.95 (photos) | ZDNet

iPhone Asterisk project is a port of the Asterisk PBX to the Apple iPhone.

iPhone Asterisk project is a port of the Asterisk PBX to the Apple iPhone. It is still in early testing, but it complies, runs, and handles calls without issue. I am working on submitting my asterisk changes to Digium, but until then I've put a binary and source release below. To get the source release to complile, you need a working iphone and have the iphone cross complier working.

The first port of Asterisk 1.4.13 is done and working. All the code is released under the GPL v2.

Tested with iPhone firmware 1.0.2, not tested with 1.1.1.

How do I install Asterisk on my iPhone?

1) Make a new dir on your iphone (/usr/local/asterisk)

2) Copy all the files from the tarball into that directory

3) Execute /usr/local/asterisk/sbin/asterisk

4) Connect to the cli: /usr/local/asterisk/sbin/asterisk -r

5) Enjoy!


Download Here

Microsoft launches open source Outlook tool, SDK projects

Microsoft appears to be serious about making Outlook more accessible to open source developers.
On May 24, the Redmond, Wash software giant announced two new open source projects designed to complement its recently released technical documentation for Microsoft Outlook Personal Folders (.pst).

The two open source projects — dubbed .pst Data Structure View Tool and .pst File Format Software Development Kit — will make it easier for developers to access data stored in digital formats created by Microsoft Outlook and use that data in cross platform solutions.

Here’s how Microsoft described the benefit of the tool and SDK, which will simplify extracting the .pst data:
Developers can use these resources to more easily build solutions, including competitive products, that run on top of the .pst file format, unlocking data stored in .pst files in simple scenarios, such as extracting photos stored in .pst files to create an album, as well as more complex scenarios, including archive search, e-discovery and corporate compliance, and uploading data to the cloud.

In the spring, Microsoft released technical documentation for Office Outlook that makes it easier for developers to read and write data out of .pst files on any platform — whether or not Microsoft Outlook is installed on the desktop.

Previously, developers could use the Messaging API (MAPI) or Outlook Object Model to retrieve and use the e-mail, calendar and attachment data but it required Outlook to be on the desktop.

How Mobile Communication works?

Acronyms

Abis - Interface as E1 line
BSC - Base Station Controller
BTS - Base Transceiver System
EIR - Equipment Identify Register
HLR - Home Location Register
MS - Mobile Station
MSC - Mobile Switching Center
PCU - Packet Control Unit
Um - Air Interface
VLR - Visitor Location Register

Indian 3G auction closes, seven of nine bidders grab concessions

The oft-delayed sale of Indian 3G spectrum on Thursday, 20 May 2010 has finally been concluded, the Department of Telecommunications (DoT) has announced, revealing that the auction raised more than double the expected amount for the government. After 34 days of bidding, seven of the nine operators that had successfully applied to take part in the sale process came away with 3G licences, although no single operator was able to lay claim to a pan-India concession, with the most that any cellco managed to win being frequencies in 13 of the country’s 22 telecoms circles. Know More...

TV meets web. Web meets TV.

Google TV is a new experience made for television that combines the TV you know and love with the freedom and power of the Internet. Watch an overview video below, sign up for updates, and learn more about how to develop for Google TV. Coming Soon..

Gmail - Account recovery via SMS: Basics

f you forget your password, you can reset it using a recovery code which we send in a text message to your mobile phone. To receive this message, you'll first need to enable account recovery via SMS on your Google Account. Know More...

Hotmail's new security features vs Gmail's old security features

Microsoft’s revamped Hotmail, set to be rolled out in mid-summer according to the company’s press release, introduces several new security features, among which are full-session SSL, visual indication for trusted email senders, and improved password recovery mechanisms. Know More...

Asterisk 1.8 will make a difference

Asterisk has added a lot of new features and internal scalability and stability since 1.4. The 1.6.x releases are to me test releases to show and run practical tests with all of these changes. The core has changed, the API’s has changed and the internal PBX is practically new. We’ve proven scalability to over 10.000 calls on one server. We’ve proven interoperability with many, many products out there. We’ve changed the way we do development and we’ve moved Asterisk into the world of non-PSTN wideband audio. Of course, there’s a lot of more things we can do, but if we consider all of the changes since 1.4, Asterisk 1.8 LTS will be a really cool telephony toolkit.


Note: support for Asterisk 1.6.0 will cease in october 2010

SIP Session Description Codes

Session description
     v= (protocol version)
     o= (owner/creator and session identifier)
     s= (session name)
     i=* (session information)
     u=* (URI of description)
     e=* (email address)
     p=* (phone number)
     c=* (connection information - not required if included in all media)
     b=* (bandwidth information)
One or more time descriptions (see below)
     z=* (time zone adjustments)
     k=* (encryption key)
     a=* (zero or more session attribute lines)
Zero or more media descriptions (see below)
Time description
     t= (time the session is active)
     r=* (zero or more repeat times)
Media description
     m= (media name and transport address)
     i=* (media title)
     c=* (connection information - optional if included at session-level)
     b=* (bandwidth information)
     k=* (encryption key)
     a=* (zero or more media attribute lines)

SIP Methods from VoIP Info

SIP Methods

3CX VoIP blog � The Busy Lamp Field (BLF)

3CX VoIP blog � The Busy Lamp Field (BLF)

Ubuntu Login into Root prompt

Type "sudo su -" in your user prompt, so that you will be entered into root mode without prompting for root password for the first time.