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July 31, 2005
Asterisk -In Windows
This is a followup of sorts as I venture into another fun challege/test of the VOIP system of choice for the monetarily challenged. Here is the line up
Windows XP Pro
Asterisk Win32 from asteriskwin32.com
AMD 64 3200
1 Gb DDR2 RAM
650 GB SATA storage
MSI NVIDIA 6600 PCI Express
A8N-SLI Deluxe Motherboard
- NVIDIA nForce4 SLI
- PCI Express Architecture
- SATA 3Gb/s
- Dual RAID
- Dual Gigabit LAN & AI NET2
- NV Firewall
- AI NOS™
- AI Audio (8-channel Audio)
My goal is to get all of this up and running with asterisk and see what the performance ramifications of running it under CYGwin are. I will use the majority of the configurations from the other ASterisk servers I have with the main difference being the Windows thing... I will post update this port in a few hours and explain the results...
Update 1:
Server is up and running with only a few minor issues. Issue 1 was broadvoice. The setup is not as they have explained in the help files they have on there site or any of the true unix implementations I have done. Issue 2 was the way it is handling the Cisco 7960 SIP phones. Currently, in incoming calls only, I can hear the other person when they call but they cannot here me. In outbound calls it is fine. The workaround I have found it to use a m switch in the extensions.cfg so that the incoming caller hears music instead of ringing. Audio is then passed throughout the entire call.
Overall the server seems to run very well. The calls are routed efficiently and the broadvoice service again is shining as a very easy way to establish calls to non VOIP equipment or phones.
I will post again the details of the change to the first issue. Currently I have modified the registration information for the broadvoice service to create teh connection/registration to them. Once I am sure it is solid I will post the sip.cfg file that succeeds.
So far Asterisk for windows is a success and the system can still be used for other tasks easily. Memory usage is low and will 10 VOIP to VOIP calls going on with translation to another codec the processor doesn't break a sweat. I am very pleased so far. Tomorrows post will bring the guts out of this config so that you can duplicate it at home without having to have a seperate box for it.
Posted by Sean at 05:26 PM | Comments (0) | TrackBack
July 29, 2005
Random Thought on website hits and weather patterns
As I was sitting here looking out my window wondering why visitors have been down for one of my other sites it dawned on me. Maybe it is not raining all over the world, just here... I am originally from Denver, Colorado and rain there is like gold. It just doesn't rain everyday. Lately where I am now it seems to rain everyday and sometimes three or four times a day. So as the counts are lower I am realizing that everybody else is probably out biking, hiking, running, or just genuinely enjoying themselfs and I am listing to the rain. My thought now is can I correlate the visitors and the times they visit to the weather patterns and predict (by knowing the weather forcast) how many people from a particular region will visit the site and how many are just going out to enjoy the day. And that brought me to another random thought. When it is 100+ degrees outside do I get more visitors then too??? This is extremely random but it now has my mind wandering to figure out how I can gauge these things. Does anywone else have this problem? Would anyone else admit to thinking this way...probably not... Anyway it was what I was sitting here thinking... Happy Friday!
Posted by Sean at 09:16 AM | Comments (0) | TrackBack
SIP 101 - Session Initiation Protocol Explained
For all of us that use it but didn't understand exactly what it was doing... This is an excellent primer on SIP. Now if I could just get my hands on one of those SIP WiFi phones...
Session Initiation Protocol or SIP refers specifically to a language that various computers can communicate to one another in so that they can complete voice calls. It has become vitally important in recent years as it plays a central role in VoIP or Voice Over Internet Protocol. VoIP Is the rapidly growing technology which has millions of Americans throwing out their local and long-distance telephone bills and replacing them with free calls made over the internet.
While Session Initiation Protocol sounds like technobabble, it helps if you can imagine SIP as the common language that new generation operators use to complete calls over the internet. With SIP, however, the operators are no longer hundreds of people in a room somewhere connecting one call to another but simply your computer device connecting to the telephone or computer device of the person you want to talk to. The fact that there is no need for real operators, or even a central board to complete calls through, explains part of why SIP is so revolutionary.
SIP was intended to give ordinary callers like you and me all the familiar functions and features of what we expect from a phone call, such as a dial tone, a ringing sound, etc. So while all the communication from our end seemed exactly the same as before, SIP makes phone calls by communicating directly with the other person’s telephone device. Unlike traditional telephony, which was based on a cog and wheel approach in which the call you placed goes through a central location and then is routed to the person you are trying to call, SIP is based on internet protocol. This means that there is no need for a central cog to run calls through, but rather calls can be made directly from person to person.
The fact that the technology is based on internet protocol (IP) rather than a traditional cog and wheel also means that placing and receiving calls are no longer inhibited by location. To conceive of this more easily it is best to think of something like your e-mail. You can take your laptop and access your e-mail from your home, just as easily as you can plug that laptop into the internet at access your e-mail from anywhere. In the same way, you can plug your SIP phone into any access point in the internet and call a person who can be located anywhere in the world. Similarly you can receive phone calls from anywhere in the world no matter where you are, simply be plugging your SIP phone into the internet.
As you can probably imagine, this ability has some pretty remarkable applications. For companies or businessmen that work out of the office, moving your office phone number is as simple as picking up the phone and carrying it with you. There is nothing else to it. For those often staying in hotels for travel or business, this means always having the direct office line with you wherever and whenever you want. No need to forward calls or even to ever pay for long distance or hotel phone access again!
The SIP technology is already revolutionizing the way in which humans communicate. In recent years, literally millions of Americans have tossed aside their traditional land based phone lines and opted for the freedom and cheapness of VoIP. Empowered with SIP technology it is uniquely able to provide you with virtually free calling, anywhere in the world, anytime in the world, without having to forward calls, change your number, or rely on others to check important messages. It is truly a technology for the future of business as well as the future of communication.
Posted by Sean at 08:14 AM | Comments (0) | TrackBack
July 28, 2005
Broadvoice & Asterisk Exceeding my VOIP needs
VOIP is starting to change my world. With Asterisk at the helm and a Cisco 7940 at the house and a 7960 on my desk at work at between my wife and myself we now have a Bat phone of sorts... She pickes it up and hits a key and mine rings...I pick up the phone and push a button and I reach her. Then if either of us need to call long distance or international... It is as simple as just dialing the coutry code and the number...Voila. And I heard that Broadvoice was a bummer of a service. I have had to call them a few times and that has been a pain, however, I have found a secret number that gets an answer every time! And the person on the other end has completely diagnosed/solved my problems. As far as I am concerned Broadvoice is much better than most of the other providers in that they have there BYOD policy which meets the needs of the advanced techno-geeks out there... I am included in that bunch. Anyway, So far Broadvoice will continue to be my provider....
Posted by Sean at 01:58 PM | Comments (1) | TrackBack
Empirix and Digium (Asterisk) Partner
I found this interesting... VOIP a sign of the times...
Empirix and Digium (of Asterisk fame) have announced that Empirix has been named as "a premier participant in the Digium Partner Program", as well as Digium's sole partner for VoIP testing. Digium, Inc., the leader in open source PBX solutions, utilizes the Empirix Hammer FX and Hammer Call Analyzer to test its current open source products, including the Asterisk Business Edition product.
"We're delighted to have Empirix as our test system partner. Hammer systems are central to the comprehensive test program that ensures Asterisk's reliability, performance, and interoperability with key hardware, software, and protocols, "said Jim Webster, director of software technologies for Digium. "We use Hammer to test Digium hardware for full compatibility with Asterisk Business Edition, as well as several select models of our open source VoIP devices. Our test bed systems are also subjected to extreme stress conditions, using Hammer test equipment, to simulate hundreds of complex VoIP calls in various real-world combinations and configurations."
"Empirix has worked closely with Digium to develop and implement test methodologies that meet the challenges of open source system development," said Duane Sword, vice president of product marketing for Empirix. "We look forward to extending that relationship through the Digium Partner Program, and to working with Digium on future product testing."
Posted by Sean at 01:42 PM | Comments (0)
VOIP
Another successful day of VOIP. Using my Cisco 7960 G phone at my house with a server located at work I am able to use four digit extensions through two firewalls and I am not having any problems. I expected this to be a huge hurdle and yet I have managed to pass through it without much fuss. The configuration is pretty basic. I am using Broadvoice as a carrier for all long distance and past my internal netowrk phone calls and a windows XP machine for the internal calls. Asterisk seems to shine here. The configuration was easy to implement.
Sip.cfg
added the broadvoice config
Exten.cfg
Added the dialing plan for the internal and external calling
and voila....
I have been calling many friends that otherwise I would have been paying LD for. I have even been able to call a friend in Ireland for next to nothing.... More to come...
Posted by Sean at 05:00 AM | Comments (0)
Asterisk Up and running
Well, I have gotten asterisk up and running... Made my first VOIP international call today! Incredible how easy it was to do. I downloaded the software and set it up with 2 cisco 7960 and and 2 cisco 7940 voip phones. It was almost too easy, it was really that simple. There is more to come as I figure out the config details. Summary of configuration:
4 Cisco VOIP phones
1 Asterisk server running Windows Media center edition
1 broadband voip provider
configured over 4 states and so far 200+ test calls made...
Testing is going great.
More to come...See tomorrow.
Posted by Sean at 02:10 AM | Comments (0)
July 27, 2005
VOIP -Explained Part 3
You pick up the receiver, which sends a signal to the ATA.
The ATA receives the signal and sends a dial tone. This lets you know that you have a connection to the Internet.
You dial the phone number of the party you wish to talk to. The tones are converted by the ATA into digital data and temporarily stored.
VoIP Terms
The central call processor is a piece of hardware running a specialized database/mapping program called a soft switch. See the "Soft Switches" section to learn more.
The phone number data is sent in the form of a request to your VoIP company's call processor. The call processor checks it to ensure that it is in a valid format.
The call processor determines to whom to map the phone number. In mapping, the phone number is translated to an IP address (more on this later). The soft switch connects the two devices on either end of the call. On the other end, a signal is sent to your friend's ATA, telling it to ask the connected phone to ring.
Once your friend picks up the phone, a session is established between your computer and your friend's computer. This means that each system knows to expect packets of data from the other system. In the middle, the normal Internet infrastructure handles the call as if it were e-mail or a Web page. Each system must use the same protocol to communicate. The systems implement two channels, one for each direction, as part of the session.
You talk for a period of time. During the conversation, your system and your friend's system transmit packets back and forth when there is data to be sent. The ATAs at each end translate these packets as they are received and convert them to the analog audio signal that you hear. Your ATA also keeps the circuit open between itself and your analog phone while it forwards packets to and from the IP host at the other end.
You finish talking and hang up the receiver.
When you hang up, the circuit is closed between your phone and the ATA.
The ATA sends a signal to the soft switch connecting the call, terminating the session. Probably one of the most compelling advantages of packet switching is that data networks already understand the technology. By migrating to this technology, telephone networks immediately gain the ability to communicate the way computers do.
It will still be at least a decade before communications companies can make the full switch over to VoIP. As with all emerging technologies, there are certain hurdles that have to be overcome.
Posted by Sean at 03:58 AM | Comments (0)
VOIP and E911
I just got an email from Broadvoice stating that I was NOT to call 911 from any of my VOIP phones useing there service. Seems like they are having a little trouble getting their E911 working. Anyway it brought me to this read...
The FCC announced a 30 day extension of its e911 rules. The FCC has stated that it will not initiate any action until August 30, 2005 against any provider of VoIP service. (This pertains to the requirement that the provider obtain acknowledgement by every customer.) The email states that it is conditional that the VoIP provider file a detailed report with the Commission by August 10, 2005.
Posted by Sean at 03:52 AM | Comments (0) | TrackBack
July 26, 2005
VOIP -Explained Part 2
Try it Yourself
If you're interested in trying VoIP, then you should check out some of the free VoIP software available on the Net. You should be able to download and set it up in about three to five minutes. Get a friend to download the software, too, and you can start tinkering with VoIP to get a feel for how it works. One place to look is http://www.skype.com.
But chances are good you are already making VoIP calls any time you place a long-distance call. Phone companies use VoIP to streamline their networks. By routing thousands of phone calls through a circuit switch and into an IP gateway, they can seriously reduce the bandwidth they're using for the long haul. Once the call is received by a gateway on the other side of the call, it is decompressed, reassembled and routed to a local circuit switch.
Although it will take some time, you can be sure that eventually all of the current circuit-switched networks will be replaced with packet-switching technology (more on packet switching and circuit switching later). IP telephony just makes sense, in terms of both economics and infrastructure requirements. More and more businesses are installing VoIP systems, and the technology will continue to grow in popularity as it makes its way into our homes.
VoIP Features
The Forrester Research Group predicts that nearly 5 million U.S. households will have VoIP phone service by the end of 2006. Perhaps the biggest draws to VoIP for the home users that are making the switch are price and flexibility.
Flexibility
With VoIP, you can make a call from anywhere you have broadband connectivity. Since the IP phones or ATAs broadcast their info over the Internet, they can be administered by the provider anywhere there is a connection. So business travelers can take their phones or ATAs with them on trips and always have access to their home phone. Another alternative is the softphone. A softphone is client software that loads the VoIP service onto your desktop or laptop. The Vonage softphone has an interface on your screen that looks like a traditional telephone. As long as you have a headset/microphone, you can place calls from your laptop anywhere in the broadband-connected world.
Price Most VoIP companies are offering minute-rate plans structured like cell phone bills for as little as $30 per month. On the higher end, some offer unlimited plans for $79. With the elimination of unregulated charges and the suite of free features that are included with these plans, it can be quite a savings.
Most VoIP companies provide the features that normal phone companies charge extra for when they are added to your service plan. VoIP includes:
Caller ID
Call waiting
Call transfer
Repeat dial
Return call
Three-way calling
There are also advanced call-filtering options available from some carriers. These features use caller ID information to allow you make a choice about how calls from a particular number are handled. You can:
Forward the call to a particular number
Send the call directly to voicemail
Give the caller a busy signal
Play a "not-in-service" message
Send the caller to a funny rejection hotline
With many VoIP services, you can also check voicemail via the Web or attach messages to an e-mail that is sent to your computer or handheld. Not all VoIP services offer all of the features above. Prices and services vary, so if you're interested, it's best to do a little shopping. Now that we've looked at VoIP in a general sense, let's look more closely at the components that make the system work. In order to understand how VoIP really works and why it's an improvement over the traditional phone system, it helps to first understand how a traditional phone system works.
The VoIP Phone System: Packet Switching
Data networks do not use circuit switching. Your Internet connection would be a lot slower if it maintained a constant connection to the Web page you were viewing at any given time. Instead, data networks simply send and retrieve data as you need it. And, instead of routing the data over a dedicated line, the data packets flow through a chaotic network along thousands of possible paths. This is called packet switching. While circuit switching keeps the connection open and constant, packet switching opens a brief connection -- just long enough to send a small chunk of data, called a packet, from one system to another. It works like this:
The sending computer chops data into small packets, with an address on each one telling the network devices where to send them.
Inside of each packet is a payload. The payload is a piece of the e-mail, a music file or whatever type of file is being transmitted inside the packet.
The sending computer sends the packet to a nearby router and forgets about it. The nearby router send the packet to another router that is closer to the recipient computer. That router sends the packet along to another, even closer router, and so on.
When the receiving computer finally gets the packets (which may have all taken completely different paths to get there), it uses instructions contained within the packets to reassemble the data into its original state.
Packet switching is very efficient. It lets the network route the packets along the least congested and cheapest lines. It also frees up the two computers communicating with each other so that they can accept information from other computers, as well.
The Advantage
VoIP technology uses the Internet's packet-switching capabilities to provide phone service. VoIP has several advantages over circuit switching. For example, packet switching allows several telephone calls to occupy the amount of space occupied by only one in a circuit-switched network. Using PSTN, that 10-minute phone call we talked about earlier consumed 10 full minutes of transmission time at a cost of 128 Kbps. With VoIP, that same call may have occupied only 3.5 minutes of transmission time at a cost of 64 Kbps, leaving another 64 Kbps free for that 3.5 minutes, plus an additional 128 Kbps for the remaining 6.5 minutes. Based on this simple estimate, another three or four calls could easily fit into the space used by a single call under the conventional system. And this example doesn't even factor in the use of data compression, which further reduces the size of each call.
Let's say that you and your friend both have service through a VoIP provider. You both have your analog phones hooked up to the service-provided ATAs. Let's take another look at that typical telephone call, but this time using VoIP over a packet-switched network:
Posted by Sean at 03:58 AM | Comments (0)
July 25, 2005
VOIP Explained -Part 1
If you've never heard of VoIP, get ready to change the way you think about long-distance phone calls. VoIP, or Voice over Internet Protocol, is a method for taking analog audio signals, like the kind you hear when you talk on the phone, and turning them into digital data that can be transmitted over the Internet.
How is this useful? VoIP can turn a standard Internet connection into a way to place free phone calls. The practical upshot of this is that by using some of the free VoIP software that is available to make Internet phone calls, you are bypassing the phone company (and its charges) entirely.
VoIP is a revolutionary technology that has the potential to completely rework the world's phone systems. VoIP providers like Vonage have already been around for a little while and are growing steadily. Major carriers like AT&T are already setting up VoIP calling plans in several markets around the United States, and the FCC is looking seriously at the potential ramifications of VoIP service.
Above all else, VoIP is basically a clever "reinvention of the wheel." In this article, HowStuffWorks will show you the principles behind VoIP, its applications and the potential of this emerging technology, which will more than likely one day replace the traditional phone system entirely.
The interesting thing about VoIP is that there is not just one way to place a call. There are three different "flavors" of VoIP service in common use today: ATA - The simplest and most common way is through the use of a device called an ATA (analog telephone adaptor). The ATA allows you to connect a standard phone to your computer or your Internet connection for use with VoIP. The ATA is an analog-to-digital converter. It takes the analog signal from your traditional phone and converts it into digital data for transmission over the Internet. Providers like Vonage and AT&T CallVantage are bundling ATAs free with their service. You simply crack the ATA out of the box, plug the cable from your phone that would normally go in the wall socket into the ATA, and you're ready to make VoIP calls. Some ATAs may ship with additional software that is loaded onto the host computer to configure it; but in any case, it is a very straightforward setup.
IP Phones - These specialized phones look just like normal phones with a handset, cradle and buttons. But instead of having the standard RJ-11 phone connectors, IP phones have an RJ-45 Ethernet connector. IP phones connect directly to your router and have all the hardware and software necessary right onboard to handle the IP call. Soon, Wi-Fi IP phones will be available, allowing subscribing callers to make VoIP calls from any Wi-Fi hot spot.
Computer-to-computer - This is certainly the easiest way to use VoIP. You don't even have to pay for long-distance calls. There are several companies offering free or very low-cost software that you can use for this type of VoIP. All you need is the software, a microphone, speakers, a sound card and an Internet connection, preferably a fast one like you would get through a cable or DSL modem. Except for your normal monthly ISP fee, there is usually no charge for computer-to-computer calls, no matter the distance.
Part 2 Coming soon
Posted by Sean at 03:56 AM | Comments (0)
July 20, 2005
What do ya know...
After all of this anoying pain of trying to get a and b to work together ... The blog server is up and running. What a joy to finaly have a place to express this headache where the wife won't say ..."Yes, and"
Posted by Sean at 03:19 AM | Comments (1) | TrackBack
July 10, 2005
Asterisk for Windows? Absolutely
Now the playing field has truly been leveled... I found this to be a good read:
Microsoft Windows™ users who want to deploy the popular Asterisk open source PBX system finally have a solution — sort of.
N2Net, a provider of mission critical communication hosting, announce the immediate availability of AstWind, a package allowing users of several Windows platform to run Asterisk, but only as a Voice over IP application.
"AstWind allows Windows users to safely install and test the Asterisk PBX with the click of a button." said Gregory Boehnlein, Vice President of N2Net. "Prospective users can test the software and become familiar with it before deploying a full Linux system. Developers can maintain multiple environments for testing and PBX Administrators can create disaster recovery plans to backup existing installations."
Most VoIP only features are available on the Windows version, although there is currently no support for any hardware interfaces. The AstWind package contains a full development environment and the Asterisk PBX pre-installed on a Debian GNU Linux filesystem.
Through the coLinux networking subsystem, Asterisk has access to the network and can participate fully as a unique server.
Cooperative Linux (short-named coLinux) is the first working free and open source method for optimally running Linux on Microsoft Windows natively. It is a port of the Linux kernel that allows it to run cooperatively alongside another operating system on a single machine. For instance, it allows one to freely run Linux on Windows 2000/XP, without using a commercial PC virtualization software such as VMware, in a way which is much more optimal than using any general purpose PC virtualization software.
"The creation of AstWind was remarkably easy, thanks to the amazing work of the coLinux project which allowed us to setup a Debian GNU Linux environment on Windows with minimal overhead," Boehnlein said. "The entire process took less than an hour. For obvious reasons, I do not advocate using AstWind for mission critical installations, but it does open some interesting doors for hobbyists and developers. With AstWind, users can experiment with the Asterisk PBX in a no-risk manner."
The AstWind installer can be downloaded as a single executable file from N2Net's high performance FTP cluster located at ftp://ftp.nacs.net/asterisk/astwind.
Posted by Sean at 01:29 PM | Comments (0) | TrackBack