Asterisk Help
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Note: Another great installment from the Nerd Vittle's Asterisk Factory.
It's birthday week at Nerd Vittles, and today you get the party favor as we introduce the second generation of our free turnkey (aka preconfigured) Asterisk system: nv-TrixBox-1.1.2. As with the first version, it runs on the desktop of any Windows XP home or office computer. If you want a state-of-the-art phone system, look no further. Out of the box, it supports eight extensions and two lines with integrated voicemail and immediate email delivery of your incoming voicemail messages.
To add additional extensions takes about 5 seconds. This PBX features the latest build of Asterisk (1.2.12.1) and is just the ticket for a small business or a school or even a fraternity or sorority house. It's also perfectly suited for your home.
You get every imaginable PBX telephony feature including music on hold, call forwarding, and call transfer as well as a preconfigured AutoAttendant which lets your friends and colleagues direct an incoming call to any of your extensions or even your cellphone. For those with the magic password, you can even dial in and get dialtone to make five hours of free calls each week to dozens of countries around the world ...
Click Here for the Full Nerd
Written by Dal on September 29th, 2006 with no comments.
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For those that thought we’d dropped off the face of the planet, good news. Not yet. If you haven’t heard, there’s a new version of TrixBox, 1.2. And we’ve given it the old college try for a week or two with about that same results pictured in this old comic book. On some platforms, it runs just fine. On others, including our VMware for Windows machines, it’s a nightmare.
The voice synthesis system is again broken, freePBX can’t reload Asterisk without completely shutting down and restarting Asterisk (amportal restart). And there appear to be all sorts of interrupt or timing problems that we’ve never seen before … going back to Asterisk@Home 1.2.
We attribute many of the problems to a new version of CentOS and Asterisk, both of which are bundled into the TrixBox 1.2 package, but who knows. What we do know is TrixBox 1.2 is a little too Bleeding Edge for our taste, and most of the Nerd Vittles goodies that depend upon the Flite speech engine no longer work on many machines...
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Written by Dal on September 25th, 2006 with no comments.
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Note: I have to say it. "This one time in Asterisk Camp". Anyways, grats to Jared and the Sokol crew for helping bring this together. This is exactly what any emerging technology needs if it is to really take hold and go main stream. Heck if I had an extra $3500.00 I would sign-up right now.
Big Nerd Ranch, Inc. announced today the newest addition to its line-up of classes, Asterisk Bootcamp, November 6-10, 2006. Asterisk Bootcamp, taught by instructor Jared Smith and developed by Sokol & Associates, highlights one of the most talked about emerging technologies in the telephony and programming industries. Asterisk is the perfect solution for a spectrum of telephony needs from voice over IP (VoIP), teleconferencing and switchboard functionality.
As the global marketplace demands greater flexibility for meetings, conferencing, and sophisticated voice systems, Asterisk, the open source contender, provides a flexible, economical and scaleable design environment ideal for cutting-edge business, military and governmental powerhouses. This bootcamp is ideal for developers, Linux/Unix consultants, interconnect vendors and ITSP system administrators looking for an enhanced and more elegant PBX solution that rises above its more pedestrian competitors.
The innovative bootcamp highlights the instruction prowess of Asterisk guru and author, Jared Smith. Jared is the Senior Consultant and instructor for Sokol & Associates and the creator of the IAX2 trunking technique. He is featured as a co-author of the seminal Asterisk text, Asterisk: The Future of Telephony, and one of the co-creators of the Asterisk Docs project.
"When you're dealing with emerging technologies, finding the right instructor with the requisite experience and street credentials to teach a class can be challenging," said Aaron Hillegass, Big Nerd Ranch founder. "Having someone with the knowledge base, reputation, and vigor of Jared Smith is like winning the lottery. His finger has been at the pulse of Asterisk design almost since its inception, and his ability to translate his knowledge into practical instruction for students is an incredible asset."
The Asterisk Bootcamp is a five day course which covers in-depth the keys points of Asterisk installation, configuration and administration. The course begins with an in-depth review of Asterisk as an application, a project and a community. It goes on to teach the student to download, compile, install and tune Asterisk; to connect it with both end-user devices (phones) and the outside world. Students will learn to create dialplans, to implement applications, and to make use of the long list of features which are included with Asterisk. At the end of the course, students should be able to create a working Asterisk system from a standard Linux computer, configure the system to support end users connected via multiple technologies, and to handle all necessary adds, moves and changes. If you are wanting to learn how to implement a working Asterisk system, this class is for you.
The bootcamp assumes previous experience with programming; especially with Linux.
Read more about Asterisk Bootcamp (including the complete syllabus) or our instructor Jared Smith.
The Big Nerd Ranch incorporates intensive training classes for Unix and Mac OS X programmers in a retreat setting outside Atlanta, GA. Class price of $3500 includes lodging, all meals, original instruction materials, 24-hour lab access, and transportation to and from the Atlanta airport. Students are encouraged to bring independent projects to class, allowing for input from classmates and individual instructor attention. For more information, call 678-595-6773 or visit:
www.bignerdranch.com
Written by Dal on August 31st, 2006 with no comments.
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Excerpt: For those that have been chomping at the bit to play with podcasting, today's your lucky day. We're going to show you how to do anything you've ever wanted to do with podcasts using your plain old telephone. All you need is a shiny new TrixBox server or any Asterisk server with freePBX installed...
Out of the box, there are two things you can do with your TrixBox system and GabCast. First, you can create podcasts. And, second you can host conference calls. Today Nerd Vittles adds the third leg to the stool. With the new GabCast Player for Asterisk, you'll be able to listen to any GabCast channel feed using a garden-variety, touchtone phone.
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Written by Dal on August 29th, 2006 with no comments.
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Editor's Note: This was sent in from one of our readers Kerry Garrison with Tech Data Pros. It's a interesting read. Please subimit any questions or comments about this article.
Excerpt: You've played around with Asterisk and set yourself up a test system and everything works but you go into production with a TDM card connected to the phone lines and now the users are complaining about echo. There wasn't any echo on the old phone system or hooking up an analog phone, and your SIP trunks sound fine, so what is the problem?
The actual number of things that can cause echo and how to resolve them all could almost fill a book, instead we are going to look at the most basic echo canceling using the Zaptel drivers. This will solve "most" echo problems but will not work in every case, regardless it is the first tier approach to solving to problem. The procedures outlined in this article will work with Digium TDM analog or TE PRI cards.
Click Here for the Full Article
Written by Dal on August 21st, 2006 with no comments.
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Excerpt: Well, it's back-to-school time, and today we introduce the first free turnkey (aka preconfigured) Asterisk system. And it runs on the desktop of any Windows XP home or office computer. If you want a state-of-the-art phone system, look no further. Out of the box, it supports eight extensions and two lines with integrated voicemail and immediate email delivery of your incoming voicemail messages.
To add additional extensions takes about 5 seconds. This PBX is just the ticket for a small business or a school or even a fraternity or sorority house. It's also perfectly suited for your home. You get every imaginable PBX telephony feature including music on hold, call forwarding, and call transfer as well as a preconfigured AutoAttendant which lets your friends and colleagues direct an incoming call to any of your extensions or even your cellphone.
For those with the magic password, you can even dial in and get dialtone to make five hours of free calls each week to dozens of countries around the world including all of the U.S. and Canada, most of Europe, South and Central America, Australia and all your Far East favorites including China, Taiwan, Russia, and Japan. And the total cost: about $12.50 for each three months of service.
All incoming calls are free, and you even get your very own area code and phone number to pass out to your friends that are still chained to plain old telephones or cellphones. And, yes, all your favorite Nerd Vittles applications are preinstalled and ready to go including weather forecasts for 1,000 airports, MailCall for Asterisk to read you your email messages, NewsClips for Asterisk to read you the news, and the AsteriDex robodialer complete with a web interface to place your outbound calls and to serve up customized CallerID for your incoming calls.
Last but not least, you get all of the bundled TrixBox applications including freePBX, SugarCRM, Samba for Windows networking, FTP and SSH support, WebMin, PHP, MySQL, Perl, Apache, SendMail, integrated fax-to-email support, calling card billing, and more. It slices, it dices ... You get the idea.
Click Here for the Full Nerd
Written by Dal on August 17th, 2006 with no comments.
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Just a quick note that Edvina in cooperation with Digium is starting the fall season of trainings again.
Coming trainings are:* Asterisk Bootcamp, Boston - next week!
We still have a few seats available
* Asterisk Beachcamp, Malaga, Spain
A class in a beach hotel in beautiful Malaga on the Spanish south coast
Both classes are bootcamp-level classes with dCAP oppurtunities.
Visit our web site for more information or send e-mail to:
info@edvina.net
Asterisk@von - Voice On The Net in Boston
---------------------------------------------
For those of you going to Von Boston there will be a series of Asterisk seminars at von, labelled Asterisk@von. I will be covering the coming release, 1.4 as well as run developer meetings and a meeting for the Asterisk Video Task Force. See the von web site at http://www.von.com for more information.
Regards,
/Olle
Written by Dal on August 16th, 2006 with no comments.
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Today we introduce the third generation of our Asterisk database dialer utility, AsteriDex. It's a web-based AutoDialer on Steroids. It lets you store and manage phone numbers of all your friends and business associates with an easy-to-use MySQL database.
Call up your contacts using your favorite web browser from anywhere and click on the contact you want to call. Then, presto! AsteriDex first calls you and then connects you to your contact through an outbound call made using your Asterisk server.
AsteriDex also can automatically look up CallerID Names in your MySQL database for incoming calls. And, it can be used as a garden-variety speed dialer from any telephone by simply spelling up to five characters of any contact's name. It'll even tell you who is being called. And, yes, it's FREE!
Click Here for the Full Nerd
Written by Dal on August 8th, 2006 with no comments.
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Well, we're on a roll adding new tricks to our
TrixBox. In the last month, we've added
Weather Reports for 1,000 U.S. Airports. And last week, we taught our TrixBox system to read email over the telephone with
MailCall for Asterisk. Today, we add another 'Speak-And-Spell' application: an RSS newsreader for your phone. With today's free software and any phone that can connect to your Asterisk system, you get access to all of the
RSS News Feeds from Yahoo.
There are dozens available with more to come. The news feeds include the latest headlines, sports, health, technology, show biz, politics, business news, and many more. NewsClips for Asterisk joins dozens of other RSS newsreaders with a couple of important differences. First, of course, our app is FREE! And second, you can use a Plain Old Telephone to get your one-minute news fix whenever you like. And the feeds are always current. Yahoo!
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Written by Dal on July 25th, 2006 with no comments.
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Last week we learned how to connect an Asterisk server to legacy phones and phone service. Today we're going to set up a connection to the outside world and set up internal extensions, so we can actually place and receive calls through Asterisk.
Starting the TDM04B at boot We need to finish configuring the TDM04B. Last week we left off with manually loading the drivers for the TDM04B, just to make sure they would. If there are errors in /etc/zaptel.conf the drivers won't load. Getting the drivers to load at boot is easy.
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Written by Dal on July 24th, 2006 with no comments.
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For those that served in the military, there are at least two things you'll probably never forget: the lousy food and mail call. We don't have a solution for the lousy food, but we've got a terrific enhancement for mail call.
We've named it MailCall for Asterisk, and it joins dozens of other telephony applications named MailCall with one important difference. Ours is FREE! What do it do? Well, it reads your email messages to you over the telephone. All you have to do is dial up your Asterisk server from any touchtone telephone. Can it handle multiple email accounts? Absolutely.
Do the email accounts have to be on the Asterisk server? Nope. Does it work with POP3 and IMAP mail accounts? Yep. Which email messages can it speak? We've tried it successfully with messages from Yahoo, and HotMail, and Google Mail, and Comcast Mail, and RoadRunner, and Outlook Express, and Notes Mail, and Entourage.
And it works with plain text messages as well as those with attachments although it doesn't deal with the attachments. No, it can't tell you what kind of picture is lurking in your inbox. Maybe someday. If you happen to be running a current version of TrixBox, then deploying MailCall for Asterisk will take you about 15 minutes.
Click Here for the Full Nerd
Written by Dal on July 19th, 2006 with no comments.
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There are thousands of people that operate their own Asterisk based PBX systems, yet they do not enable any method to allow for p2p sip URI dialing. These sip "targets" are very easy to enable and allow you to dial anyone that has also enabled the function.
Dialing with SIP URI completely avoids toll calling and forces your Asterisk server to create P2P sip connections when you dial someone's SIP URI. It makes a less complex phone call without a system administrator configuring a peer and best of all: It gets rid of phone numbers and your telco!
How does it work?
By creating a SRV record in DNS for your domain you can help remote PBX systems establish P2P calls for a specific extensions. For example, when someone calls me, my URI is resolved to my PBX (sip.blyon.com). When the call comes into my Asterisk box, blyon is setup as a extension, and that extension is connected to a phone or a context. As a result, if someone uses something like Xten to call blyon@blyon.com, I get a normal ring and phone call. When I use my Cisco 7960 phone and dial someone's SIP URI it completes like a normal phone call.
Why is this cool?
This is great because it takes away any central control for locating people. The ENUM standard is nice, but gives someone else control over the mapping database and it keeps an ugly old phone numbers in place. I really don't want to dial phone numbers 10 years from now, I much rather just give someone my email address and have that map to my phone. If I need to call a business, I much rather just call pbx@somecompany.com then find some obscure phone number.
If more people adopt this as a standard, it will be the method of choice for calling people and it puts power into the end user's hands!
Click Here for the Full Article
Written by Dal on July 13th, 2006 with no comments.
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Today we'll show you how to install the latest and greatest TrixBox 1.1 with freePBX 2.1.1 in just over an hour. As with the earlier release of TrixBox, these new Asterisk products are designed to support the casual home or home office user's PBX needs as well as gigantic call centers processing millions of calls a month.
Everything is free except the hardware on which to run your new system. That can be almost any old Pentium PC or a multi-processor RAID box with mainframe horsepower. We also want to get TrixBox properly configured to support our next free application: TrixBox MailCall.
It'll let you retrieve and play back your email messages using any touchtone telephone and your TrixBox 1.1 system. And, yes, you'll need TrixBox 1.1 to make everything work.
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Written by Dal on July 10th, 2006 with no comments.
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It's fun and useful to explore pure IP telephony, but most folks want PSTN (Public Switched Telephone Network) integration as well. (PSTN is also known as POTS: Plain Old Telephone Service.) An Asterisk server can interface with "legacy" systems, which means "not IP," or your ordinary old digital and analog telephone systems.
Digium Inc., the inventors and sponsors of Asterisk, make legacy interface hardware. Other popular brands are Sangoma, VoiceTronix, and Cisco. There are dozens of different vendors, as most networking hardware manufacturers have added VoIP devices to their product lines. I'll use Digium's products as examples. These require the Zaptel drivers, which you can download from asterisk.org, and the Linux operating system—because the Zaptel drivers only work on Linux.
Other operating systems can use standalone media gateways. "Media gateway" is a broad term that includes both devices for servers, and devices for individual telephones. You can use a media gateway with your Linux system, rather than installing interface cards in your Asterisk server. Analog interfaces An Asterisk server plugs nicely into an existing analog phone network.
Just add an adapter like Digium's TDM2400P. This connects to your existing punch block with a standard 25-pair telco cable and connector. Now you have a powerhouse PBX that can do just about anything, for a fraction of the cost of a traditional PBX. The TDM2400P cards cost from around $600 to $1,700, depending on how they are configured. You may use the TDM2400P to connect your existing analog phones to your VoIP network, which allows you to replace them with IP phones on a timetable that suits you. Or never replace them, whatever fits into your master plan.
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Written by Dal on July 10th, 2006 with no comments.
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One of the more fun aspects of running an Asterisk server is choosing your own hold music. Hopefully your callers are not sitting on hold for long periods of time, but as long as they are it's nice to give them something pleasant to listen to. It can be music, inspirational speeches, comedy routines—even Internet radio.
Linux comes with all the tools you need to record sound files and convert them to the Asterisk-friendly .gsm format. Audacity is an excellent sound recorder and editor that runs on Linux, Macintosh, and Windows. As fun as it is, we're not getting into sound recording today, though, just managing existing sound files.
Asterisk versions 1.2 and later includes their own music player, so you can ignore all the documentation that tells you how to add one. Asterisk can handle several different sound file formats on its own, but decoding and encoding sound files eats up CPU cycles, so it's more efficient to convert them yourself.
Written by Dal on July 7th, 2006 with no comments.
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The Cisco 7970 probably has the best voice quality of any telephone we've ever used. And we've used lots of them. But the Hobson's Choice for most folks is this. Do you want great sounding IP phone calls with a phone that costs two to five times as much as other IP phones while giving up virtually every other feature that has made IP telephony great?
While it will let you retrieve your voicemail messages from your Asterisk server, unfortunately you'll never know you have a message unless you dial in regularly and manually check. This phone has been pitched as the perfect phone for the busy executive. The first busy executive that misses an important meeting because the message waiting lamp never lit up, and this phone would be out the window. Too bad!
Written by Dal on January 1st, 1970 with no comments.
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This is a reminder that the Twin Cities Asterisk Users Group will be meeting this Saturday, October 14th at 10:30am. - Please note the time change; we are meeting one hour earlier than our normal time. This month is our bi-annual new user meeting. We'll show you how to get started with Asterisk and answer your questions about what Asterisk can do and if possible, we'll show you how, on the spot.
The new Asterisk 1.4 is currently in the beta test stage and if there is interest, we'll update one of our systems from 1.2 to 1.4 and discuss what's new and what changes you might need to make. If you're not a developer, this is now your chance to contribute to the Asterisk development process. Beta testers are needed now. Please try the new version on any non mission critical systems.
Meetings are held monthly on the second Saturday of each month, excluding July and December. The Agenda is posted online
http://www.voip-info.org/wiki/index.php?page=Twin+Cities+Asterisk+User+Group+Agenda
Meetings are held at Sound Choice Communications LLC... -= 7839 12th Ave So, Bloomington Minnesota USA 55425 =-
http://maps.google.com/maps?oi=map&q=7839%2012th%20Ave%20S%2055425
Come to a meeting to meet other asterisk users, see asterisk solutions, win a door prize, eat food, or for the good company, to look for work, if your looking for employees, to go out for a drive, to get out of your house, whatever, JUST COME TO THE MEETING!
New visitors can help themselves to FREE FXO Interface cards (So you can connect your phone line, and have a timing source for meetme and IAX protocols). Some members have been known to swap hardware at the meetings. Have extra VoIP gear, looking for VoIP gear? There's plenty of hardware to see. Have you been to a meeting recently?
Please come and share your own ideas and learn from others. As always, free food.
We are always looking for help with meeting topics. If you feel like taking the lead, please do and simply let me know if you need anything.
Meeting starts at 10:30am and parking is available in the rear of the building. Runs about 2 hours or less, and we'll order Pizza to the meeting for lunch. This month we will need to wrap up by 12:30pm or 12:45pm.
Look forward to seeing you there.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20User%20Group% 20TwinCities%20Minnesota%20USA
If you have a product or service you'd like to introduce to our members, send a private message to ejo1(at)soundchoicecomm.com and we'll see if we can't get you listed as next month's sponsor.
Sound Choice Communications is a reseller of Digium and Polycom products and we have inventory on hand. Give us a call and your items will be waiting for you on Saturday. Thank you! +1.651-999-0888
Written by Dal on January 1st, 1970 with no comments.
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Note: Nice to see the Vittle's crew released a VMWare Image. I love being about to boot up Fedora Core 5 anytime its needed.
Today we'll show you how to install the latest and greatest TrixBox 1.2.3 in about an hour. It is by far the best Asterisk-based IP PBX on the planet... especially once you add all of the Nerd Vittles goodies. It's been a painful couple of months in the TrixBox community, but the wait is over. Whether you're a casual home user or a gigantic call center processing millions of calls a month, this IP PBX can do it all reliably.
The best news: everything is FREE except the hardware on which to run your new system. That can be almost any old Pentium PC or a multi-processor RAID box with mainframe horsepower. HINT: There's even a hidden link to the upcoming VMware image of TrixBox 1.2.3 for your Windows Desktop which has the whole Nerd Vittles enchilada preloaded.
Written by Dal on January 1st, 1970 with no comments.
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NetLogic, a leading provider of Business-Class VoIP (
voice over Internet protocol) and Internet services, today announced the creation of its "PBX-Preferred" Certification program. NetLogic's initial PBX-Preferred Certifications recognize Asterisk-based software and systems; hence the Company announced its new program in conjunction with
AstriCon 2006, the Asterisk Conference and Exhibition in Dallas, October 24 - 27, 2006.
Written by Dal on January 1st, 1970 with no comments.
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Excerpt: It's almost Halloween at Nerd Vittles, and today you get a real treat as we introduce the third generation of the free turnkey Asterisk system for Windows: nv-TrixBox-1.2.3. With a few minor changes, this version is about as rock-solid as any Asterisk system on the planet. Of course, the planets do continue to move so be sure to check back here from time to time and review all the newly posted comments...
As with the prior versions, it runs on the desktop of any Windows XP home or office computer. If you want a state-of-the-art phone system, look no further. Out of the box, it supports eight extensions and two lines with integrated voicemail and immediate email delivery of your incoming voicemail messages. To add additional extensions takes about 5 seconds.
And, yes, all your favorite Nerd Vittles applications are preinstalled and ready to go including weather forecasts for 1,000 airports, MailCall for Asterisk to read you your email messages, NewsClips for Asterisk to read you the news, the AsteriDex robodialer complete with a web interface to place your outbound calls and to serve up customized CallerIDs for your incoming calls, TeleYapper to broadcast reminders and messages to your clients or little league team, and our new GabCast (podcasting) Player for Asterisk.
Click Here for the Full Nerd
Written by Dal on January 1st, 1970 with no comments.
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Note: This is a great idea. I am sigining up for sure. If this gets off the ground I will add an "Asterisk Enterprise" section to host questions and answers. Great idea Anthony.
Greetings,
This is my annual post-Astricon attempt to get an Enterprise Asterisk User Group off the ground. We are a municipal government using Asterisk to replace a legacy PBX. I'd be interested in starting a
group of similar enterprise users (
say, 100 seats or more) other than resellers, carriers and call-centers who are using Asterisk to support their non-telecom-related business - I don't envisage any geographical limitation to the group (there seem to be few enough of us as it is!).
If you are interested, please let me know off-list.
Regards,
--
Anthony Rodgers (CunningPike)
Business Systems Analyst
District of North Vancouver
Web:
http://www.dnv.org
Written by Dal on January 1st, 1970 with no comments.
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Lots of talk about VoIP these days - and those who have been following know that we implemented
Switchvox - a VoIP PBX in our office over a year ago.
In general - it has gone well. Switchvox is based on
Asterisk - and there were some bugs that seemed to conflict with our PRI hardware. It's still not clear to me where the problem was. Brian @ Switchvox worked pretty hard to patch SV for us to bypass the problem once he isolated it. June's update, SV has been stable and we've been happy with it's maintainability and flexibility.
've wanted to be able to use it to connect with Skype and/or gtalk .. and yesterday I finally it working pretty well with gtalk - so thought I'd document it a bit here in case others want to do the same. It's certainly possible to do this in Asterisk as well - but it's likely that most Switchvox users are not so sophisticated as the traditional Asterisk or Trixbox users .. so here's the overview of how to do it in Switchvox. I may be adding SipBroker here unnecessarily - but I don't have Switchvox set up to accept SIP calls directly - so I like having Sipbroker in the middle to handle that for me and to simplify dialing - as you'll see below..
What you need:
Go get 'em now. I'll wait for you. On the gtalk2voip account - don't bother getting a "business" account. A personal one is fine.
At sipbroker - you need to set up your Switchvox with sipbroker as a provider. This will make it much easier for you to dial out to gtalk - and in from PSTN (regular) phones.
Written by Dal on January 1st, 1970 with no comments.
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Note: This came across the list today. Some helpful tutorials.
Hello list,
I have prepared a couple of new tutorials you may find interesting:
- Installing an Asterisk 1.4 beta system - at: http://astrecipes.net/?n=216
- Installing the Digium's Asterisk GUI for 1.4 - at: http://astrecipes.net/?n=217
It's nothing too complex, but you may find them interesting, especially the new Asterisk GUI.
Any comment is welcome - the site is a wiki, so feel free to correct any errors or add improvements.
l.
--
Loway Research - Home of QueueMetrics
http://queuemetrics.loway.it
Written by Dal on January 1st, 1970 with no comments.
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Excerpt: If you're as lazy as the rest of us, then getting up to change what's playing on iTunes or to adjust the volume is just too much like work especially if you've installed one of our PBX-in-a-Flash Asterisk systems on either a dedicated Linux machine or your Windows Desktop.
For long-time readers of Nerd Vittles, you may recall that we covered how to build a streaming audio server using iTunes last year. So today we add the missing piece which will let you change songs, adjust the volume, and pause and restart iTunes using any touchtone phone connected to your Asterisk or TrixBox system.
Written by Dal on January 1st, 1970 with no comments.
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Note: Good old Mundy knows when to give us the right application at the right time of year. Kudos
Excerpt: After we published our article providing Asterisk weather forecasts for 1,000 U.S. cities using airport codes, we heard from a number of folks asking if we could do something similar using zip codes to retrieve forecasts from the National Weather Service. Well, Santa is two weeks early, and today you get your wish: an enhanced weather application for Asterisk that supports every last zip code in the United States, all 42,740 of them.
As with our original weather application based upon airport codes, this one requires the Flite voice synthesizer which now is an integral part of all TrixBox builds. You'll note that there are no thermometers and barometers to install with this application.
So, to be perfectly candid, this is really a virtual weather station. The National Weather Service does the forecasting, the U.S. Postal Service does the locating, and the Asterisk Weather Station simply provides the glue to put the two together and retrieve and play the results using your touchtone telephone.
The good news is that you actually control and manage this application rather than worrying about dialing into someone else's system and finding that they've gone out of business or quit providing the service.
Written by Dal on January 1st, 1970 with no comments.
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Note: Shawn Van Every posted this on the list. This makes me really happy to see. If Asterisk wants to really hits the big leagues, acedemic support will go a long ways imho.
I wanted to drop a line as I just finished up teaching a class that used Asterisk extensively and I wanted to pass along major kudos to all have contributed to Asterisk as well as to point out some of the technical difficulties that I ran into.
I posted the to following to my blog:
academic asterisk adventures
In my continuing adventures as an adjunct professor at NYU’s Interactive Telecommunications Program, I taught a new course last semester entitled “Redial: Interactive Telephony“.
The purpose of the course was to utilize emerging telephony technologies, concepts and services such as VoIP as a tool for building interactive applications and devices. ITP students are
famous for their imaginative use of new technology. One of my goals in this course was to help them apply some of their creative and critical thinking to new telephony technology in the context of the rich history of telephony.
The course content was focused around voice and touch-tone based applications using Asterisk, SIP, RTP, text to speech (using Festival), speech recognition (using Sphinx) and the like.
The results of this course were truely fantastic and I will take a bit of time in the coming days/weeks to highlight their projects.
For a quick taste, checkout these projects: http://itp.nyu.edu/show/results.php?searchstring=Redial&submit=Search
For now I want to write a bit about using Asterisk in an environment such as this in the hopes that some Asterisk users/developers offer suggestions and perhaps implement some of my suggestions.
We (18+ students in my class, myself and a couple of students following along without actually being members of the class) used an older Intel P3 machine running RedHat Enterprise Linux 4 (as I recall) with the Secure Linux enabled. We used Asterisk 1.2 and each student had a normal linux user account on the machine. We also had Apache 2, PHP, MySQL (running on another machine), PHP AGI, Perl, Festival and Sphinx.
Many of the issues that we ran into were a direct result of running PHP for both AGI scripting in Asterisk and normal web development with Apache. PHP’s safe mode and SE linux contributed to these difficulties.
The problem is that Asterisk was running as the “asterisk” user and Apache was running as a different user. PHP safe mode was relaxed to allow the group to execute the files but this still threw many of us for a loop several times.
I don’t want to whine about this too much as it is what was available and we just had to deal with it. One thing that might help would be to enable suexec within Asterisk so that AGI scripts could run as the owner of the script. This, I believe is how Apache can be setup to handle things and would go a long way towards alleviating many of the issues we had with both security and usability.
The next major problem we had was in developing dialplans and editing other Asterisk configuration files. At first, I setup an extensions file for each student that was included (using #include) in the main Asterisk extensions.conf file. Unfortunately, we ran up against a hard limit to the number of includes that Asterisk would handle and half of the files never got included.
To alleviate that problem, I put together a PHP script and a shell script that would cat together all of the extension files that needed to be included. These scripts also took care of issuing the reload
command to the asterisk manager interface. This worked reasonably well but didn’t have any error checking so that if one user’s extensions file had errors or if they used a context that was named
the same as another user problems would arise.
This is probably a harder problem to solve in the current design of Asterisk. I am interested in hearing other’s thoughts on how these problem could be solved. My thoughts are that Asterisk could somehow take some pointers from Apache and allow individual users to have a set of configuration files that get included at run time when their extension is entered. Something similar to the concept of a public_html directory. Asterisk when told to go to a specific user’s context would look in a specific directory and include the dialplan from there.
Perhaps I am just dreaming.. ;-) What do you think?
In any case, none of this would have been possible in a world without Asterisk and on behalf of my students and myself a big thank you to all those who have contributed to Asterisk!
-SVE
Written by Dal on January 1st, 1970 with no comments.
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Note: Matt Birkland, a local
Seattle Asterisk PBX Integrator sent in this article after working through some troubles installing ABE on the Fedora Core 5 Operating System. I hope this helps out anyone that has been having trouble. Please email (comments[at]asteriskvoipnews.com) in any comments or questions and we will be sure to pass them along and make updates as needed.
Full Article:
Many people may be in a situation like myself where they need specific driver support for a raid array for their Asterisk server. In my case this requires me to use
Fedora core 5.
Fedora Core 3 and
4 is supported by Digium, but not
5. However I called Digium to confer and a tech told me that they will attempt to support any issue they can replicate on a supported system.
They will not support the installation itself. Luckily I figured it out, this procedure will likely work on Fedora Core 6 as well, but I have yet to confirm this.
The Zaptel drivers compile fine(get your 2.6-dev installed to be safe) the main problem is that ABE looks for specific libraries of older versions found in Fedora Core 3 and 4. Below is the list of failed dependencies and how to install the correct ones without breaking half the programs.
NOTE:
Ignore config.h errors. The config.h file is usually found in the kernel headers more commonly used in distributions with earlier versions of the 2.6 kernel. ABE will compile Zaptel just fine without it, but some driver modules, like my laptops wireless rtl8185 card will not compile without config.h (found in: /usr/src/kernels/ 2.6.18-1.xxxx.fc6-i686/include/linux/config.h). Just add a blank text file called 'config.h' to get around most of these issues.
Asterisk Business Edition compiles with these errors on Fedora Core 5:
libcrypto.so.4 is needed by asterisk-BusinessEdition-B.1_2.i686
libnewt.so.0.51 is needed by asterisk-BusinessEdition-B.1_2.i686
libodbc.so.1 is needed by asterisk-BusinessEdition-B.1_2.i686
libssl.so.4 is needed by asterisk-BusinessEdition-B.1_2.i686
Follow these three steps to get complete install:
1) install libssl-0.7a to meet the libssl.so.4 and libcrypto.so.4 requirement.
WARNING: Uninstalling the current openssl with the Add/Remove Programs could break your system (I reinstalled once, many programs depend on this). The idea is we want to add the new library without disturbing the programs that use the new one.
a. yum upgrade openssl
b. yum install openssl-0.9a
2) install the newt library. Fedora Core 5 and 6 use version newt-0.52. The Fedora yum repository does not have 0.51 or a compatabilty package, this will not install unless you force it. You don't need libslang, just force it. Download the package from a popular rpm site. Like freshrpms.net or rpmfind.net.
a. rpm --force -hiv libnewt-0.51
3) install unixODBC compat library
add/remove software or use yum to install unixODBC compat.
Thats it! Run the install script again, register with Digium and complete installation.
Matt Birkland
Network Engineer
VoiceIP Solutions
Written by Dal on January 1st, 1970 with no comments.
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Note: Found this handy help article on how to setup a approx. $150 GSM gateway from ebay to work with your Asterisk PBX. Internally on our office we are going to be deploying a GSM gateway to get a couple employees' cellphones without the big bills and the ability to test this concept out.
"The GSM gateway MV-370 is manufactured by http://www.portech.com.tw/.
It's main interest in comparison with other gateways is that it is GSM to SIP (not FXS), so voice quality is very good.
Another point of interest is the price (around $150 on ebay).
With that gateway properly configured, you are able to receive calls from GSM to Asterisk (including DISA) and to give calls from Asterisk to GSM network.
"
Usage
A typical usage of such a gateway is to be able to give a call with your normal mobile to any destination at voip cost :
Your mobile <----gsm network----> MV-370 <--lan--> Asterisk <--internet--> VOIP provider <--whatever--> landline
To do such a call, you just call your MV-370 number (it has its own simcard), then you get an invitation tone, then you dial the number which is handled by Asterisk.
If you have some special deals with your mobile operator, like free special number, you can call your MV-370 for free. You can then call all around the world from your mobile at voip cost :-)
Written by Dal on January 1st, 1970 with no comments.
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Excerpt: Thanks to the work of literally hundreds of developers, there is a terrific Asterisk IP PBX with an incredible array of additional bells and whistles. That product which we have tested extensively is TrixBox 1.2.3. It’s so good, in fact, that we chose it as the base system for all of the Nerd Vittles applications that we write about each week.
What was missing unfortunately was a way to run this same system on a Mac. So today we have not one but two special treats for the Mac enthusiasts of the world.
First, it’s now possible to run our standard Version 3 system using the new VMware beta for the Intel Mac. And, thanks to one of our great contributors, there’s now another alternative: a Parallels image of our Version 3 Asterisk system. Today, we’ll show you how to install both of them…
Written by Dal on January 1st, 1970 with no comments.
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Note: Matt Birkland (Network Engineer) is on a roll by sending in another Asterisk Help article about Email Name Resolution on internal networks.
One of the most popular features in Asterisk is the Voice Mail to email feature. It's fairly straight forward to set up as far as Asterisk is concerned, but many people (including myself) have run intoo a situation where the Asterisk server is on the same internal network as the mail server. When Asterisk resolves somebody@somecompany.com , it attempts to send the mail to the Fully Qualified Domain Name (FQDN).
In the example above this might be something like 216.39.144.X. This works great if you have hosted email outside your network, or you have internal DNS to send the mail orginating inside the network to the correct host. However many small to medium size businesses eithe don't have internal DNS or don't know how to set it up correctly.
The result is that the VM attachment is being sent outside the network and can't get back because of a routing loop. This discussion isn't about routing loops, just trust me that sending out packets out of your network when the destination is on the inside causes problems. If you want more go get your CCNA. Moving on...
Asterisk by default uses sendmail to send the VM attachments. Sendmail will by default resolve from it's name server BEFORE it resolves from the /etc/hosts file. Luckily, I have a solution that may just work for you.
The first step is to install the sendmail-cf package. Without this package we cannot reconfigure sendmail.
Example:
[root@localhost ~]# yum install sendmail-cf
Altenativly you can use the add/remove software program in Fedora Core or whatever your distribution of choice is.
The second step is to edit /etc/mail/sendmail.mc and rebuild sendmail. In the example below I'm using the nano text editor, but any will do. Note: I'm only showing the portion of we need to edit. Before we make any changes lets back up the sendmail.mc file. Type: 'cp sendmail.mc sendmail.mc.old'. Great job, we're ready to go. Follow the example below.
Example:
[root@localhost ~]# nano /etc/mail/sendmail.mc
sendmail.cf:
------------------------------------------------------------------------------------
dnl # Uncomment and edit the following line if your outgoing mail needs to
dnl # be sent out through an external mail server:
dnl #
dnl # define(`SMART_HOST',`smtp.your.provider')
dnl #
-----------------------------------------------------------------------------------
Now change the SMART_HOST name to something arbitrary like 'mymailserver' and uncomment it.
Example:
---------------------------------------------------------------------------------------
dnl # Uncomment and edit the following line if your outgoing mail needs to
dnl # be sent out through an external mail server:
dnl #
define(`SMART_HOST',`mymailserver')
dnl #
------------------------------------------------------------------------------------------
Now we have to update these changes to sendmail and restart the server.
Example:
[root@localhost mail]# make -C /etc/mail
Followed by a restart of the service.
[root@localhost ~]# service sendmail reload
reloading sendmail: [ OK ]
reloading sm-client: [ OK ]
[root@localhost ~]#
Great, if you made it this far you are either a huge nerd or your job is on the line. Either way we're almost done. The last step is to make an entry in your /etc/hosts file.
Example:
/etc/hosts
-------------------------------------------------------------------
# Do not remove the following line, or various programs
# that require network functionality will fail.
127.0.0.1 localhost.localdomain localhost
192.168.1.18 mattserv.voiceipsolutions.com
#IP Address of mail server below
192.168.1.200 mymailserver
-------------------------------------------------------------------
Thats it! Call and leave yourself a message.
Source:
Written by Dal on January 1st, 1970 with no comments.
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Last year, I took a trip to Asia. To stay in touch, I carried a GSM world phone, capable of receiving telephone calls in the countries I was visiting. The capability to receive calls with the same mobile phone number I use at home while halfway across the world seemed incredibly cool-at least until the first call came in!
Mobile phones hide the location of the phone, which cuts both ways. A colleague had decided to call me in the middle of the day on a Friday, which had awakened me very early on Saturday morning, because the phone "hid"my faraway location from him.
After returning home, I asked several people why my phone company could not simply play a message to warn callers when my time zone changes by more than four or five hours, letting them know the call might be inconvenient. Nobody could come up with a technical reason, but we all suspected it was because the mobile phone company to which I subscribed charged several dollars per minute to connect calls.
As part of the process of attaching a GSM phone to a network, the home network needs to learn where the phone is visiting, and that information conceivably could include a time zone. I returned to my idea once I started using Asterisk, because it provides an extensive toolkit for designing PBX-hosted services. Anything that can be coded in a computer can become an Asterisk service.
After I understood the basics of Asterisk, I sat down to implement a feature that kept track of the time of day where I visited and prevented calls from coming in at inconvenient times. The system I built on top of Asterisk to handle this feature has two major parts. The key to the system is maintaining a time-zone offset from the time in London.
(My code implements offsets only of whole hours, though it could be extended to use either half or quarter hours.) When a device first connects to Asterisk, its IP address is used to guess the location and, therefore, the time offset. After the offset is programmed into the system, incoming calls are then checked against the time at the remote location. Before the phone is allowed to ring, the time at the remote location is checked, and callers can be warned if they are trying to complete a call at an inconvenient time.
Written by Dal on January 1st, 1970 with no comments.
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The one constant in every Asterisk Integrators life is the pain of dealing with telco's. The main problem with telco's is the disparity between the tech support.
Sometimes you open a ticket with the guy that can detect an issue in hours. Usually it takes days of bitching and finger pointing to actually get someone on site. Many times a Telco insist their PRI/T1's are funtioning, but your Asterisk server is dropping frames. As a consequence, popping, cracking, lost calls, static and echo occur. This is a really bad place to be, if you spent hours trying to convince your boss about the merits of Asterisk.
Then after weeks of argument the telco's admit some kind of 'crosstalk' on the line and they dispatched a Qwest technician to fix it. This happens so often that I don't even get mad anymore, it's like a bad joke affecting an entire Industry. However... as I learned recently, sometimes symptoms that look like a telco issue are actually IRQ issues.
Most Asterisk Integrators are aware that IRQ issues are huge problem and take steps to secure their Digium 110P (
T1) or TDM40B (
analog lines). However on most modern OS's (
like Linux) IRQ's are handled logically by the kernel through APIC (
Advanced Programmable Interrupt Controller). The great thing about APIC is that you can have multiple interrupt controllers to deliver all the IRQ's necessary for running todays boundless devices. The downside is that the output from 'cat /proc/interrupts' is not always the same as scanning the PCI bus with the 'lspci -bv' command. So while it may seem like your devices are on separate IRQ's, in some cases they are not. Lets look at the example below.
[root@localhost asterisk]# cat /proc/interrupts
CPU0
0: 44302787 IO-APIC-edge timer
7: 0 IO-APIC-edge parport0
8: 1 IO-APIC-edge rtc
9: 1 IO-APIC-level acpi
14: 113862 IO-APIC-edge libata
15: 1588284 IO-APIC-edge ide1
50: 0 IO-APIC-level uhci_hcd:usb3
58: 1904417 PCI-MSI eth0
169: 0 IO-APIC-level uhci_hcd:usb4
225: 177207785 IO-APIC-level uhci_hcd:usb2, wcfxo
233: 0 IO-APIC-level uhci_hcd:usb1, ehci_hcd:usb5
NMI: 1
LOC: 44304635
ERR: 0
MIS: 0
Seems pretty straight forward; this is how I have been checking IRQ's for years. Now take a look from lspci command below.
Note: This is not the full output. I only cut and paste the Digium card and the PCI devices it shares.
[root@localhost asterisk]# lspci -bv
0a:02.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface
Subsystem: Unknown device 8085:0003
Flags: bus master, medium devsel, latency 32, IRQ 10
I/O ports at 6000
Memory at d8300000 (32-bit, non-prefetchable)
Capabilities: [40] Power Management version 2
0a:03.0 VGA compatible controller: ATI Technologies Inc ES1000 (rev 01) (prog-if 00 [VGA])
Subsystem: Giga-byte Technology Unknown device 515e
Flags: bus master, stepping, medium devsel, latency 66, IRQ 10
Memory at d0000000 (32-bit, prefetchable)
I/O ports at 6400
Memory at d8310000 (32-bit, non-prefetchable)
Capabilities: [50] Power Management version 2
05:00.0 Ethernet controller: Intel Corporation 82573V Gigabit Ethernet Controller (Copper) (rev 03)
Subsystem: Giga-byte Technology Unknown device 108b
Flags: bus master, fast devsel, latency 0, IRQ 10
Memory at d8200000 (32-bit, non-prefetchable)
I/O ports at 5000
Capabilities: [c8] Power Management version 2
Capabilities: [d0] Message Signalled Interrupts: 64bit+ Queue=0/0 Enable-
Capabilities: [e0] Express Endpoint IRQ 0
00:1f.3 SMBus: Intel Corporation 82801G (ICH7 Family) SMBus Controller (rev 01)
Subsystem: Giga-byte Technology Unknown device 27da
Flags: medium devsel, IRQ 10
I/O ports at 3080
In short, if you don't have an IRQ reserved at the BIOS level remember to scan your PCI bus with the 'lspci -bv' command. With this many devices on the same IRQ all sorts of errors and strange behavior is possible. If the IRQ is not saved for that specific device the best course of action to switch the PCI slot your Digium card resides in.
Matt Birkland
Network Engineer
Seattle Business Phone Systems
Written by Dal on January 1st, 1970 with no comments.
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