Category: Asterisk

Link: Using IP tables to secure Linux server against common TCP hack attempts

 

Important
This is an edited version of a post that originally appeared on a blog called The Michigan Telephone Blog, which was written by a friend before he decided to stop blogging. It is reposted with his permission. Comments dated before the year 2013 were originally posted to his blog.

This article was originally published in November, 2010.

Iptables
Image by Jordan W via Flickr

I’m not entirely certain of the original source of this article — I found it on one site, but a quick search reveals that the original source is most likely this site, but I may be wrong. The author of that article says he took some of the info in that article (looks like more than “some” from where I sit) from this article: How to: Linux Iptables block common attacks

Related articles found on that site are Using iptables to secure a Linux based Asterisk installation against hack attempts and Securing Asterisk – Fail2Ban (and that latter article looks suspiciously similar to this one: Fail2Ban (with iptables) And Asterisk).

I don’t know how valid or useful any of this is, but if you are running iptables on your system (if you’re not sure enter iptables -V on the command line — it should show you the version of iptables that is installed, if it is installed) then you might want to check these articles out.  And if you find an earlier source for any of these, let me know and I’ll include the links.  I know that in the technical community sometimes information gets copied around, but would it kill you guys to give attribution and a link to the original source when you are lifting information (or even raw text) from someone else’s article?

Link: Using FreeSWITCH to add Google Voice to Asterisk

 

Important
This is an edited version of a post that originally appeared on a blog called The Michigan Telephone Blog, which was written by a friend before he decided to stop blogging. It is reposted with his permission. Comments dated before the year 2013 were originally posted to his blog.

EDIT (2018): This article is extremely out-of-date and in no way useful today, and will probably be removed from this site at some point in the future. You might find this article more useful: How to use Google Voice with FreePBX and Asterisk without using XMPP or buying new hardware.

For those of you using Asterisk, Bill over at the PSU VoIP blog has come up with a way to interface Asterisk with Google Voice, by co-installing FreeSWITCH (which also supports Google Voice).  Turns out that Asterisk and FreeSWITCH can co-exist on the same server, though you do have to change the configuration a bit so they don’t compete for the same ports.  Anyway, Bill has come up with a how-to on adding Google Voice integration to current versions of Asterisk, so if that interests you, head on over and have a look:

Using FreeSWITCH to add Google Voice to Asterisk

The bonus is that once you get FreeSWITCH installed you can play around with it and look at some of its other features, if you are so inclined. Of course, the Asterisk folks could backport the Google Voice support to previous versions and make it unnecessary to do things like this, but I’m not holding my breath.

EDIT (January 26, 2012): The Google Voice channel drivers in Asterisk 1.8 have become unreliable enough (in my personal opinion, anyway) that I just used the technique shown in this article, and I must say that it works a LOT better than Asterisk 1.8’s Google Voice support.  I also added some comments to that article (probably too many!) that among other things show how I got it working for multiple Google Voice accounts.  So I would now recommend using this method to bridge Asterisk to Google Voice in preference to using Asterisk 1.8’s native channel drivers (unless you are very short on memory and/or storage space) — it just works, and calls connect faster.  Read the article AND the comments under it first, so you’ll know what to expect, and do be aware that it takes a relatively LONG time to compile and install FreeSWITCH (compared to Asterisk).  At points during the installation it may look like it’s stuck in an endless loop, but it really isn’t. Just go away and take a walk outside or something, and come back in a while and it should be done.

How to forward a call if a remote extension is unreachable in FreePBX 2.x

 

Important
This is an edited version of a post that originally appeared on a blog called The Michigan Telephone Blog, which was written by a friend before he decided to stop blogging. It is reposted with his permission. Comments dated before the year 2013 were originally posted to his blog.

DISCLAIMER AND WARNING: This article contains EXPERIMENTAL code. DO NOT USE IT IN A PRODUCTION ENVIRONMENT until you have thoroughly tested it AND MODIFIED IT to meet your needs. It is guaranteed to NOT work (at least not in a way that will be useful to you) if you simply copy and paste it, and even if you read and follow the instructions below I don’t guarantee a thing (Asterisk can be funny, sometimes code that runs fine on one system will not on another). Anyway, you are NOT allowed to use this code unless you are willing to take ALL responsibility for modifying and testing it to make sure it will work in your situation. If you use this code and in some way it winds up not working and costing you money, don’t come after me because I’m warning you now that might happen, and it’s the risk you alone assume if you attempt to use this code!

FreePBX and Asterisk allow you to call forward a call on a busy or no-answer condition (as well as unconditionally), but there is no provision for specific forwarding if an extension (presumably an offsite one) is unreachable over the Internet. It is possible to do this, although in Asterisk 1.4 it’s not at all elegant. Some commercial VoIP providers offer a feature similar to this, calling it “Failover”, “Network Unavailable Forward” or just “Unavailable Forward”, “Network Availability Number ®” (Vonage trademarked that one!), or some similar name, but FreePBX and Asterisk do not offer similar functionality — there is no “Call Forwarding Unreachable” setting.  However, with a bit of work and a minimal amount of dialplan creation, you can emulate this feature.

Here’s an example that may work in many situations (as written it works with SIP extensions only, but maybe you can modify it slightly if you need to use it with IAX2 extensions or some other type):

First, if you are still using Asterisk 1.4 or earlier, add the following code to etc/asterisk/extensions_custom.conf:

[custom-unreachable-test] exten => _X!,1,Noop(Testing for unreachable extension ${EXTEN})
exten => _X!,n,TrySystem(asterisk -rx "sip show peers" | grep ^${EXTEN}/${EXTEN}[[:space:]] > /tmp/${EXTEN}.flag)
exten => _X!,n,ReadFile(reachable=/tmp/${EXTEN}.flag,1)
exten => _X!,n,GotoIf($["${LEN(${reachable})}" = "0"]?extoffline)
exten => _X!,n,Noop(Extension ${EXTEN} is reachable - sending to *${EXTEN} voice mailbox)
exten => _X!,n,Goto(from-internal,*${EXTEN},1)
exten => _X!,n(extoffline),Noop(Extension ${EXTEN} is NOT reachable)
;This is where you enter special forwarding conditionals for each unreachable extension
exten => _X!,n,GotoIf($["${EXTEN}" = "1101"]?from-internal,18005558355,1)
;This is the fallover (voicemail) destination in case no special destination is specified
exten => _X!,n,Noop(WARNING - no unreachable destination specified for extension ${EXTEN} - trying to send to voicemail)
exten => _X!,n,Goto(from-internal,*${EXTEN},1)

If you are using Asterisk 1.6 or later then use this instead:

[custom-unreachable-test] exten => _X!,1,Noop(Testing for unreachable extension ${EXTEN})
exten => _X!,n,Set(reachable=${SHELL(asterisk -rx "sip show peers" | grep ^${EXTEN}/${EXTEN}[[:space:]])})
exten => _X!,n,GotoIf($["${LEN(${reachable})}" = "0"]?extoffline)
exten => _X!,n,Noop(Extension ${EXTEN} is reachable - sending to *${EXTEN} voice mailbox)
;This is where you enter special forwarding conditionals for each unreachable extension
exten => _X!,n,GotoIf($["${EXTEN}" = "1101"]?from-internal,18005558355,1)
;This is the fallover (voicemail) destination in case no special destination is specified
exten => _X!,n,Noop(WARNING - no unreachable destination specified for extension ${EXTEN} - trying to send to voicemail)
exten => _X!,n,Goto(from-internal,*${EXTEN},1)

In both of the above examples, change the number 1101 to match an actual extension number on your system and change the 18005558355 to the actual number you want to send calls to (note this could be another extension on your system, including a custom extension or a ring group). Duplicate the line containing those values for each extension you may want to forward, changing those two vales in each line appropriately (also see the comment section for another possible approach).

The above code assumes that if an extension is reachable, but is busy or does not answer, you want the call to go to voicemail (* + the original extension number — obviously, this would be easy to change if that’s an incorrect assumption). However, if the extension is unreachable, you want to reroute it to the user’s cell phone or some other number. In the above example, if extension 1101 receives a call and is unreachable, it would be forwarded to TellMe at 1-800-555-TELL (18005558355) – obviously not practical in a real-world situation, but it’s just an example. Again, note you have to duplicate that line in the code for each extension that might be forwarded in this way.

In order to make this work, you need to go into the FreePBX Tools menu and select “Custom Destinations”, then add a new custom destination. The destination must be custom-unreachable-test,${EXTEN},1 and the description can be anything you want (I suggest “Unreachable Extension Test” or something similar).

For each extension you wish to use this with, you must have qualify=yes (or set qualify to a valid numeric value) in the extension settings.

Finally, for each extension you want to use this with, create a Follow-Me (or edit any existing one) for that extension. You can leave the defaults as they are (or change them if you want – maybe you want to change the Ring Time, for example) but the one thing you must change is the Destination if no answer. Change that to the Custom Destination that you just created. Also, don’t forget to add the line in extensions_custom.conf to actually do something with calls to that extension when the extension is unreachable.

The reason I say this code is not elegant is because it relies on a kludge. It does a “sip show peers”, then looks for the pattern ${EXTEN}/${EXTEN} (e.g. 1101/1101) at the start of a line, which on most systems indicates the extension is connected. This may not be the case if you are using what is known as “deviceanduser” mode (which you probably aren’t unless you’re running a call center) so in that case you may need to use a different pattern match, for example:

exten => _X!,n,TrySystem(asterisk -rx "sip show peers" | grep ^${EXTEN}[[:space:]] | grep OK > /tmp/${EXTEN}.flag)

The result of the system call will be written to the file /tmp/1101.flag (or a similar file with a different extension number) and will either contain the full line from “sip show peers” (if the device or phone is reachable) or nothing (it will be an empty file). So in the next line we read the file in (actually just one character) and test the length – if it’s zero, then that’s when we do the unreachable processing. If it’s non-zero, we send the call to voicemail. EDIT: In Asterisk 1.6 and later there’s no need to create a temporary file.

I’m not saying this is the best way to do this, or the only way to do it, but it is a way that seems to work in VERY limited testing (at least on a system running Asterisk 1.4.35 and FreePBX 2.5).

This was inspired by a thread I saw on the PBX in a Flash forum, which also notes that there may be an even better way to do this in Asterisk 1.8, but since I don’t have it and VERY few FreePBX users are running 1.8 at this time, I’m not even going to touch that one.

EDIT: I did a VERY limited test of this (and made one change in the above code as a result) on an Asterisk 1.8 system running F—PBX 2.8. There are now two code sections above, one for those running Asterisk 1.4 or earlier, and one for those running Asterisk 1.6 or later (only tested with Asterisk 1.8). And before you try that “even better way” mentioned in the last paragraph, note that “Having chan_sip set HASH(SIP_CAUSE,) on the channel carries a significant performance penalty because of the usage of the MASTER_CHANNEL() dialplan function” and that Digium has “decided to disable this feature by default in future 1.8 versions” (see this page for more information). So, probably best to stick with the method shown here, if you can get it to work for you.

Review of FreeSWITCH 1.0.6 by Anthony Minessale, Darren Schreiber, Michael S. Collins (Packt Publishing)

 

Important
This is an edited version of a post that originally appeared on a blog called The Michigan Telephone Blog, which was written by a friend before he decided to stop blogging. It is reposted with his permission. Comments dated before the year 2013 were originally posted to his blog. In order to comply with Federal Trade Commission regulations, I am disclosing that he received a free product sample of the item under review prior to writing the review, and that any links to Amazon.com in this article are affiliate links, and if you make a purchase through one of those links I will receive a small commission on the sale.
Cover of FreeSWITCH 1.0.6

In case you’ve never heard of FreeSWITCH, it is a “telephony software engine”, which means it’s in the same category as Asterisk. Over the years I’ve noticed that some Asterisk users have become frustrated with Asterisk due to unfixed bugs and design flaws that mean that the software doesn’t always work as it should. So, for quite some time, I’d hoped that a viable alternative to Asterisk might emerge, if only to keep the Asterisk developers on their toes. Competition between software projects tends to be a healthy thing, and from what I’ve read in this book, it appears that FreeSWITCH just may be the software product that eventually replaces Asterisk as the #1 open source telephony software engine.

Before I begin, as is my custom with such reviews, let’s start with a quick overview of what’s in each chapter (for the complete Table of Contents, see the Packt Publishing web site):

  • Preface
  • Chapter 1: Architecture of FreeSWITCH – includes notes on the FreeSWITCH design and important modules
  • Chapter 2: Building and Installation – how to build and run FreeSWITCH under Linux/Unix, Mac OS X, or Windows
  • Chapter 3: Test Driving the Default Configuration – here you learn how to control FreeSWITCH with the CLI and to make your first call
  • Chapter 4: SIP and the User Directory – includes adding users, setting up voicemail, and setting up a gateway to connect to the world (link is to sample chapter in PDF format at the Packt Publishing site)
  • Chapter 5: Understanding the XML Dialplan – this gets into the “meat” of FreeSWITCH dialplan creation
  • Chapter 6: Using the Built-in XML IVR Engine – here’s where you learn one way to build an IVR (auto-attendant)
  • Chapter 7: Building IVR Applications with Lua – really an example of using a scripting language with FreeSWITCH. A few other languages are supported
  • Chapter 8: Advanced Dialplan Concepts – if Chapter 5 was the hamburger, this is the sirloin
  • Chapter 9: Controlling FreeSWITCH Externally – explains the event system architecture, and how to read and send events
  • Chapter 10: Advanced Features and Further Reading – includes multi-user conferencing, billing, XML/Curl, alternative endpoints, and configuration tools and related projects

There are also two appendices:

  • Appendix A: The FreeSWITCH Online Community
  • Appendix B: The History of FreeSWITCH

The  Packt Publishing web site also has this to say about the book:

What you will learn from this book :

  • Set up a basic system to make and receive phone calls, make calls between extensions, and utilize basic PBX functionality
  • Avoid common implementation mistakes and deploy various features of this telephony system with best practices and expert tips
  • Perform routine maintenance for smooth running and troubleshoot the system when things are not going right
  • Apply regular expressions to unlock unique and powerful call routing scenarios
  • Call your own application(s) when particular events occur and control FreeSWITCH using the powerful Event Socket
  • Set up multi-party conferencing facilities for your system
  • Interact with callers, gather information, and route calls to the appropriate recipient using the automated, built-in XML IVR (Interactive Voice Response) engine
  • Create a flexible dialplan, and allow third-party tools to be quickly and easily created using dialplan parsers other than the default XML Dialplan
  • Park multiple calls in a FIFO queue and unpark them in the order in which they were received, using the mod_fifo module
  • Record an entire phone call or session using the call recording feature
  • Create advanced call control applications with the Lua scripting language
  • Take a peek into the vibrant online community and history of FreeSWITCH

Approach

This book is a step-by-step tutorial with clear instructions and screenshots to guide you through the creation of a complete, cost-effective telephony system. You will start with installation, walk through the different features, and see how to manage and maintain the system.

Who this book is written for

If you are an IT professional or enthusiast who is interested in quickly getting a powerful telephony system up and running using the free and open source application FreeSWITCH, this book is for you. Telephony experience will be helpful, but is not required.

Now, here are my impressions. Please bear in mind that I did not actually attempt to build a working FreeSWITCH installation (I would need yet another spare computer to do that), but I certainly feel as though I could after reading this book. One thing that is somewhat uncommon about this book is that the author of the software is also one of the authors of the book. Too often, when you see a book written about a piece of software, the writer doesn’t fully understand the software and therefore makes guesses and assumptions about how it works, that may lead to problems down the road if you follow their advice. When the software author collaborates on the book, that’s far less likely to happen, and indeed, at no point in this book did I get the feeling that the author was struggling to understand the subject. I will even go so far as to say that this is one of the best written technical books I have read in a long time.

The biggest complaint I had about this book — and it is a very minor one — is that it could have benefited from another proofreader. Occasionally I’d see an obvious error that the proofreader should have caught — nothing major, and nothing I couldn’t figure out with about two seconds of thought, with one exception.  On page 91 of the book, it appears to me as though there is some missing text at the bottom of the page.  It’s discussing making a test call to Music on Hold and then, suddenly and jarringly, it jumps into a time of day example.  I think the disconnect occurs in middle of a sentence: “In our example, call the debug output is as follows:”  The sentence as written does not make sense to me, and it appears a block of text (perhaps a large one) may have been omitted at this point. But that is the only place in the book where I encountered an error of that magnitude. I have submitted the error to Packt Publishing and I’m hoping they will figure out what was supposed to go there and place it in the errata section of their web site.

One other point I will make about a software author writing a book on his own creation is that I think sometimes, it’s difficult for the author to correctly envision how end users will want to use the software.  As an example, virtually all the dialplan examples in this book are in XML.  There may be advantages to using XML, but it’s not going to be very familiar to someone coming from an Asterisk background, and I might have wished for a few non-XML examples.  On pages 158-159, the author notes that,

There is a common misconception that the FreeSWITCH Dialplan is based on, and requires, XML. That is simply not true. If you prefer flat files, you could use them to store your Dialplan configuration. If you prefer YAML, you could use that, too. You just need to load the correct C-based Dialplan module to interpret your stored logic for the particular type of configuration file you want FreeSWITCH to utilize.

This aside, the most common (and currently, the most robust) Dialplan processing mechanism in FreeSWITCH is still the XML-based Dialplan module. Most Dialplan examples that are shipped with FreeSWITCH, or those scattered on the Web are in XML, therefore, they will remain the focus of this chapter. …..

Indeed, there is even an Asterisk dialplan module, albeit with limited capabilities.  From page 199:

If you are used to the Asterisk Dialplan, some basic functionality is provided by the Asterisk Dialplan module, although it is not nearly as feature-rich as the XML engine. You can process contexts and route calls to phones using the Asterisk Dialplan. This module, again, is more of a sample on how to build an alternate Dialplan processing module and should not be utilized as a full, feature-rich Dialplan system.

Yet you won’t find examples using flat files, YAML, or Asterisk Dialplan in the book.  However, the XML examples were clearly written and easy to understand, so I don’t think that there would be a steep learning curve to start writing dialplans in XML, assuming you are a proficient enough coder to write dialplans in the first place.  And, I suspect that XML would be easier for a new user to pick up than any of the other options.

I mention the above to emphasize two points:  FreeSWITCH is different from Asterisk. If you are thinking about moving from Asterisk to FreeSWITCH, you need this book to get you up to speed on the differences.  And second, FreeSWITCH is both more capable than Asterisk, and arguably easier to use, once you get used to the differences (or if you have no prior experience with similar software). FreeSWITCH appears to have been designed from the ground up to avoid the issues that have plagued Asterisk, particularly those that cause Asterisk to fall to its knees under heavy load or heavy call volumes. Even if you’re a long-time Asterisk user, you may want to get this book just to see what you’re missing.  You might decide that it’s worth your effort to set up a test system using FreeSWITCH, to help you understand how much better the next generation of telephony software engines can be.

One other point, in case you are reading this review several months after I wrote it — the author notes this in the preface:

At the time of this writing this book, the FreeSWITCH developers were putting the finishing touches on FreeSWITCH version 1.2. While the examples presented in this book were specifically tested with version 1.0.6, they have also been confirmed to work with the latest FreeSWITCH development versions that form the basis of version 1.2. Do not be concerned about the fact that this material does not cover version 1.2—it certainly does. The FreeSWITCH user interface is very stable between versions; therefore, this text will be applicable for years to come.

There will no doubt be some of you who are reading this that wonder if there are any Web GUI “front ends” (dialplan and configuration file generators) for FreeSWITCH.  Indeed there are, and they are covered in Chapter 10, which briefly explains the differences between WikiPBX, FreePBX v3, FusionPBX, and 2600hz.  Even if you plan on using a Web GUI, there may be times when you find the need to write a bit of custom code, and in that case having this book available would definitely be helpful to you.

One other thing I personally found interesting in this book was Appendix B, “The History Of FreeSWITCH.”  This explains how FreeSWITCH came to be, and along the way offers further explanation on how it is different from Asterisk and why the developers felt the need to start a new project.  What I think I found most interesting (and perhaps unfortunate, depending on your point of view) is that FreeSWITCH could have been the basis for Asterisk version 2, had only the Asterisk developers reacted positively to the idea. I see this sort of thing happen occasionally in the open source community, where the lead developers of a project start to develop an attitude that does not encourage outside contributions (or, they treat contributions or suggestions for improvement as if they were piles of steaming dog poo on their doorstep). Perhaps this should serve as a cautionary tale to such developers that your project can always be replaced by something better, if you do not encourage contributions to your own project from those not currently in your “inner circle” of developers.

As you may know if you have read my previous reviews, it’s rare that I get wildly enthusiastic about a book.  In this case I’ll make an exception, because overall the book is that well-written (my comments above notwithstanding). If you have any interest at all in using FreeSWITCH, or are even just curious about it, you really should buy this book.  It’s available in both traditional softcover dead-tree format, and as a DRM free Adobe PDF eBook, and there’s even a package deal if you want both formats. Don’t forget that you can view a sample chapter (PDF format) prior to purchase. EDIT: Also, there is an online article by the book’s authors entitled FreeSWITCH: Utilizing the Built-in IVR Engine.

FreeSWITCH 1.0.6 by Anthony Minessale, Darren Schreiber, Michael S. Collins (Amazon affiliate link)

Related: Review of FreeSWITCH Cookbook by Anthony Minessale, Michael S Collins, Darren Schreiber, Raymond Chandler (Packt Publishing)

A Perl script to send Caller ID popups from Asterisk to computers running Growl under OS X on a Mac or Growl for Windows

 

Important
This is an edited version of a post that originally appeared on a blog called The Michigan Telephone Blog, which was written by a friend before he decided to stop blogging. It is reposted with his permission. Comments dated before the year 2013 were originally posted to his blog.
[notice]EDIT March, 2014: If you are running OS X Mavericks we recommend that you do NOT use the script shown here, but instead send notifications to a XMPP/Jabber account and use iChat to receive them, since the message will then display in the Notifications Center and you do not need Growl. See How to send various types of notifications on an incoming call in FreePBX for more information. You may also find this thread on the RasPBX forum useful.[/notice]

Quite some time ago, I wrote a post explaining how you could poll a Linksys or Sipura VoIP adapter or phone once per second, and whenever there was an incoming call, generate a notification popup on your computer, if you have the Growl notification service installed.  However, that method doesn’t work if you’re not using a Linksys or Sipura phone or device.

If you are running Asterisk, there’s another way to do it, and that’s to get Asterisk to send the notifications directly. In order for this to work, the computer on which you want to receive the notifications has to be running Growl (under Mac OS X) or Growl for Windows. You must also configure Growl to receive network notifications. I will note here that if you are using a Mac and have never done that before, you may want to make sure that Growl network notifications work before proceeding, because it appears that under OS X, it’s pretty much a crap shoot whether Growl network notifications will work at all, and when they don’t the Growl folks apparently have no clue as to why they don’t. It seems to be a machine-specific thing – on some Macs they work fine, while on others they don’t work at all.

You must have the Perl language installed on your Asterisk server, and you must have the Net::Growl and Asterisk::AGI modules installed (I’m going to assume you know how to install a Perl module from the CPAN repository – if you have Webmin installed, it can be done from within Webmin). Chances are you already have Asterisk::AGI installed, unless you built your Asterisk server “from scratch” and never installed it, but if you’ve never installed Net::Growl you’ll need to do that first.

Next you want to copy and paste the following Perl script to the filename /var/lib/asterisk/agi-bin/growlsend.agi on your Asterisk server (to create a non-existent file, you can use the touch command, and after that you can edit it in Midnight Commander or by using the text editor of your choice). If this code looks somewhat familiar, it’s because it’s adapted from some code that originally appeared in a FreePBX How-To, which I modified.

#!/usr/bin/perl
use strict;
use warnings;
use Net::Growl;
use Asterisk::AGI;
my $agi = new Asterisk::AGI;
my %input = $agi->ReadParse();
my $num = $input{'callerid'};
my $name = $input{'calleridname'};
my $ext = $input{'extension'};
my $ip = $ARGV[0];

if ( $ip =~ /^([0-9a-f]{2}(:|$)){6}$/i ) {
    $ip = $agi->database_get('growlsend',uc($ip));
}

unless ( $ip =~ /^(d+).(d+).(d+).(d+)$/ ) {
    exit;
}

open STDOUT, '>/dev/null';
fork and exit;

if ( $ARGV[2] ne "" ) {
    $ext = $ARGV[2];
}

# Define months and weekdays in English

my @months = (
    "January", "February", "March", "April", "May", "June",
    "July", "August", "September", "October", "November", "December"
);
my @weekdays = (
    "Sunday", "Monday", "Tuesday", "Wednesday",
    "Thursday", "Friday", "Saturday"
);

# Construct date/time string

my (
    $sec, $min, $hour, $mday, $mon,
    $year, $wday, $yday, $isdst
) = localtime(time);
my $ampm = "AM";
if ( $hour > 12 ) {
    $ampm = "PM";
    $hour = ( $hour - 12 );
}
elsif ( $hour eq 12 ) { $ampm = "PM"; }
elsif ( $hour eq 0 ) { $hour = "12"; }
if ( $min < 10 ) { $min = "0" . $min; }
$year += 1900;

my $fulldate =
"$hour:$min $ampm on $weekdays[$wday], $months[$mon] $mday, $year";

# Next two lines normalize NANP numbers, probably not wanted outside of U.S.A./Canada/other NANP places
$num =~ s/^([2-9])(d{2})([2-9])(d{2})(d{4})$/$1$2-$3$4-$5/;
$num =~ s/^(1)([2-9])(d{2})([2-9])(d{2})(d{4})$/$1-$2$3-$4$5-$6/;

register(host => "$ip",
    application=>"Incoming Call",
    password=>"$ARGV[1]", );
notify(host => "$ip",
    application=>"Incoming Call",
    title=>"$name",
    description=>"$numnfor $extn$fulldate",
    priority=>1,
    sticky=>'True',
    password=>"$ARGV[1]",
    );

Also, if you want to be able to specify computers that you wish to send notifications to using MAC addresses rather than IP addresses (in case computers on your network get their addresses via DHCP, and therefore the IP address of the target computer can change from time to time), then you must in addition install the following Perl script. It requires a command-line utility caller arp-scan so install that if you need to – I used to use nmap for this but they changed the output format, making it harder to parse, and arp-scan is much faster anyway. Call it /var/lib/asterisk/agi-bin/gshelper.agi and note that there are two references to 192.168.0… within it that you may need to change to reflect the scope of your local network, if your network’s IP addresses don’t start with 192.168.0.:

#!/usr/bin/perl
use strict;
use warnings;
my @mac;
# Change the following lines to reflect the scope of your local network, if necessary
my @arp = `arp-scan --quiet --interface=eth0 192.168.0.0/24`;
foreach (@arp) {
        if (index($_, "192.168.0.") == 0) {
                @mac = split(" ");
                `/usr/sbin/asterisk -rx "database put growlsend \U$mac[1] $mac[0]"`;
        }
}

Make sure to modify the permissions on both scripts to make them the same as other scripts in that directory (owner and group should be asterisk, and the file should be executable), and also, if you use the gshelper script, make sure to set up a cron job to run it every so often (I would suggest once per hour, but it’s up to you).

Now go to this page and search for the paragraph starting with, “After you have created that file, check the ownership and permissions” (it’s right under a code block, just a bit more than halfway down the page) and if you are using FreePBX follow the instructions from there on out (if you are not using FreePBX then just read that section of the page so you understand how this works, and in any case ignore the top half of the page, it’s talking about a different notification system entirely).  But note that if you use the above code and have the gshelper.agi program running as a cron job, then after the first time it has run while the computer to receive the notifications is online you should be able to use a computer’s MAC address instead of the IP address.  This only works if you’ve used the modified script on this page, not the one shown in the FreePBX How-To.  As an example, instead of

exten => ****525,1,AGI(growlsend.agi,192.168.0.123,GrowlPassWord,525)

as shown in the example there, you could use

exten => ****525,1,AGI(growlsend.agi,01:23:45:AB:CD:EF,GrowlPassWord,525)

(the above is all one line) where 01:23:45:AB:CD:EF is the MAC address of the computer you want to send the notification to.  Once again, just in case you missed it the first time I said it, this won’t work until the gshelper.agi script has been run at least once while the computer to receive the notifications was online.  If for some reason it still doesn’t appear to work, run the nmap command including everything between the two backticks (`) directly from a Linux command prompt and see if it’s finding the computer (depending on the size of your network, it might be several seconds before you see any output, which is why I don’t try to run this in real time while a call is coming in).

If you are NOT running FreePBX, but instead writing your Asterisk dial plans by hand, then you will have to insert a line similar to one of the above examples into your dial plan, except that you don’t need the four asterisks (****) in front of the extension number, and if it’s not the first line in the context, you’ll probably want to use n rather than 1 for the line designator (and, you won’t be putting the line into extensions_custom.conf because you probably don’t have such a file; instead you’ll just put it right in the appropriate section of your dial plan).  In other words, something like this (using extension 525 as an example):

exten => 525,n,AGI(growlsend.agi,192.168.0.123,GrowlPassWord,525)

This line should go before the line that actually connects the call through to extension 525.  I do not write Asterisk dial plans by hand, so that’s about all the help I can give you. And if you don’t write your dial plans by hand, but you aren’t using FreePBX, then I’m afraid you’ll have to ask for help in whatever forum you use for advice on the particular software that you do use to generate dial plans, because I can’t tell you how to insert the above line (or something like it) into your dial plan.

Virtually everything in this article has already been published in one place or another, but I wanted to get it into an article with a relevant title and cut out some of the extraneous explanations and such.  There are links to all the original sources throughout the article, so feel free to follow those if you want more in-depth commentary.

Geolock — a Perl script for Asterisk or FreePBX users to enhance security

 

Important
This is an edited version of a post that originally appeared on a blog called The Michigan Telephone Blog, which was written by a friend before he decided to stop blogging. It is reposted with his permission. Comments dated before the year 2013 were originally posted to his blog.

I created the following Perl script and have been running it for a week or so, and it seems to be working well.  The idea is that this script runs once per minute, and whenever a SIP or IAX extension is registered with your Asterisk system the script looks at the IP address that the extension is registering from, and if that address is outside your home country (the United States by default), the IP address is immediately banned using IPtables.  So, your remote extensions could be anywhere in your home country and connect to your system, but if a hacker from some other nation penetrates your system and somehow guesses one of your passwords, they will (in theory) have less than a minute to do any damage before they are banned. And once they are banned, the rightful user of that extension should still have no difficulty connecting, as long as they are not coming in from outside your home country.

For those of you that don’t have any extensions connecting from outside your home country, consider this another tool in your arsenal of defenses against intrusions.  Combine it with strong passwords, and Fail2Ban with IPtables for additional security.

NOTE THAT THIS SCRIPT IS NOT GUARANTEED TO DO ANYTHING AT ALL, other than take up space on your computer.  IT SHOULD STILL BE CONSIDERED EXPERIMENTAL until there has been more testing on it.  I believe it works properly, but have no way to do extreme testing on it to see if or how it might break. THERE IS NO WARRANTY OF ANY KIND!!!

Prerequisite: Obviously, you must have IPtables installed and functioning properly, and you must install either the Geo::IP or Geo::IP::PurePerl Perl module (do NOT install both!). You can install one of these using Webmin (using Webmin’s Others | Perl modules page), or in any other way you usually install Perl modules (e.g. CPAN). The difference between the two was explained in my original article, as follows:

… there is a Perl module called Geo::IP, which calls the GeoIP C API. If you install that API (when downloading, I’d go into the test/ directory and get the latest beta) using the directions on the linked page, and then install the Perl module (you must do it in that order, or installation of the Perl module will fail), you could run a Perl script that shows the location that your off-site extensions are coming in from. If you don’t want to install the API, or can’t figure out how (not difficult if you follow the directions), you can use the Geo::IP::PurePerl Module which is slower, but does not require the additional C library. Just so you know, GeoIP puts its data file at /usr/local/share/GeoIP/GeoIPCity.dat and they suggest that you go to http://www.maxmind.com/download/geoip/database/ every month or so to grab the latest database (the full link for the country database is currently http://geolite.maxmind.com/download/geoip/database/GeoLiteCountry/GeoIP.dat.gz or you can get a much larger city-level database at http://geolite.maxmind.com/download/geoip/database/GeoLiteCity.dat.gz — just make sure you don’t grab a CSV version by mistake!). If you need them, there are installation instructions for the database, although they are primarily for the city-level database. If you buy an account, they’ll give you automatic updates, though I can’t imagine it would be that hard to write a script to do that (or maybe Google could help you find one, or see this message thread).

If the above is confusing to you, I’d stick with the Geo::IP::PurePerl Module. That should at least get you going. You also need the Perl Sys::Syslog module (unless you want to omit all the syslog-related instructions), though chances are you may already have that one.

Now, here is the Perl script. I called it geolock.pl (although you are free to name it whatever you want), and I put it in the /var/lib/asterisk/agi-bin/ directory and made it executable. Note this is in a code block so the long lines will overflow the column width, so you need to copy and paste this into a text editor. Also there are certain lines you will need to change, as explained below:

#!/usr/bin/perl
use strict;
use warnings;
use Geo::IP;
use Sys::Syslog;
my $gi = Geo::IP->new(GEOIP_STANDARD);
my ($ext, @peerline, @extension, $ipaddress, $country, $shellcmd);
my $flag = 0;

my @sippeers = `asterisk -rx "sip show peers" | grep -v 192.168.0. | grep ^[1-9] | grep -v "(Unspecified)" | grep / | sort -n`;
# change grep statements in above line to match your local IP range (1st grep) and first digits of extensions (2nd grep)
foreach (@sippeers) {
@peerline = split(" ");
$ipaddress = $peerline[1];
$country = $gi->country_code_by_name($ipaddress);
@extension = split("/",$peerline[0]);
$ext = $extension[0];
#    print "Extension $ext has IP address $ipaddress which is in country $countryn";
if ($country && $country ne 'US') {
$shellcmd = `iptables -D INPUT -p ALL -s $ipaddress -j DROP 2>&1`;
system("iptables -A INPUT -p ALL -s $ipaddress -j DROP");
openlog($0,'pid','user');
syslog('notice', "Banning IP address $ipaddress in $country because Asterisk SIP Extension $ext is connecting from there");
closelog;
if (index($shellcmd, "Bad rule") >= 0) {
$shellcmd = 'echo "This is an automated message - please do not reply. IP address ' . $ipaddress . ' in country ' . $country . ' was banned in iptables because Asterisk SIP extension ' . $ext . ' was connecting from there." | mail -s "IP address banned on Asterisk box" root@localhost';
system($shellcmd);
$flag = 1;
}
}
}

### INCLUDE THIS NEXT SECTION ONLY IF YOU HAVE IAX2 EXTENSIONS ###
my @iaxpeers = `asterisk -rx "iax2 show peers" | grep -v 192.168.0. | grep ^[1-9] | grep -v "(Unspecified)" | grep -v "iax2 peers" | sort -n`;
# change grep statements in above line to match your local IP range (1st grep) and first digits of extensions (2nd grep)
foreach (@iaxpeers) {
@peerline = split(" ");
$ipaddress = $peerline[1];
$country = $gi->country_code_by_name($ipaddress);
$ext = $peerline[0];
#    print "Extension $ext has IP address $ipaddress which is in country $countryn";
if ($country && $country ne 'US') {
$shellcmd = `iptables -D INPUT -p ALL -s $ipaddress -j DROP 2>&1`;
system("iptables -A INPUT -p ALL -s $ipaddress -j DROP");
openlog($0,'pid','user');
syslog('notice', "Banning IP address $ipaddress in $country because Asterisk IAX2 Extension $ext is connecting from there");
closelog;
if (index($shellcmd, "Bad rule") >= 0) {
$shellcmd = 'echo "This is an automated message - please do not reply. IP address ' . $ipaddress . ' in country ' . $country . ' was banned in iptables because Asterisk IAX2 extension ' . $ext . ' was connecting from there." | mail -s "IP address banned on Asterisk box"
root@localhost';
system($shellcmd);
$flag = 1;
}
}
}
### END OF SECTION ONLY NEEDED IF YOU HAVE IAX2 EXTENSIONS ###

if ($flag == 1) {
`asterisk -rx "restart now"`;
}
else {
openlog($0,'pid','user');
syslog('info', "Completed normally");
closelog;
}

These are the things you need to change for your local installation:

  • If you used the Geo::IP::PurePerl module then be sure to change the two references to Geo::IP to Geo::IP::PurePerl.
  • Remove the optional section for IAX2 extensions if you don’t have any of those (but keep it if you have any IAX2 extensions, even if they are only internal ones).
  • In the line(s) “if ($country && $country ne ‘US’) {” change US to the code for your home country if you are somewhere else in the world. You could also create a more expansive conditional statement here to allow multiple countries, or a more restrictive ones if you want to block an IP that doesn’t resolve to any country (I consider that a database error, but maybe you don’t). If you have installed the city-level database, you could even (in theory) test for something other than country, such as a state or province, city, time zone, ISP, etc. This line appears in the SIP section and also the optional IAX2 section, so make sure you change both lines if necessary.
  • There are two instances of root@localhost in the above script (in the SIP section and also the optional IAX2 section), which you should change to a valid e-mail address if you want to receive e-mail notifications when an IP address is banned. If you don’t want to receive such e-mail notifications, then comment out or remove those two lines in their entirety (they start with: $shellcmd = ‘echo “This is an automated message …) and also remove the following system($shellcmd); line(s).

The following apply to the two lines that begin with "my @sippeers” and, in the optional IAX2 section, “my @iaxpeers”:

  • Change “grep -v 192.168.0.” to a regular expression or pattern that will match it IP address of all extensions on your local network.  The pattern as shown works if all your local extensions will be in the range 192.168.0.x. Since you can use a regular expression, you could do something like “grep -v 192.168.[1-5].” which would match any local address from 192.168.1.0 through 192.168.5.255.
  • Change grep ^[1-9] to match the first digit of your extensions – as shown, anything starting with a digit 1 through 9 would be considered an extension.  The idea here is that we only want to look at extensions, not trunks (which you can restrict using permit and deny statements, if necessary). Most trunks don’t begin with a number (when you do a “sip show peers” or “iax2 show peers” listing from the CLI), so this is what separates extensions from trunks.  You may have to get a bit more creative if you have a trunk that starts with a number that overlaps your extensions. If you run the entire section between the backticks (the ` characters) from a Linux command prompt,  it should show you all of your connected non-local (that is, not on your internal network) SIP or IAX extensions (depending on which line you run), but no trunks, local extensions, offline extensions, or header information.

Sharp-eyed readers may observe that I first try to delete an iptables rule before creating it.  That’s because I don’t want to create the same rule multiple times (iptables happily accepts duplicates, unfortunately).  If I get an error when trying to delete the rule, then I know that what follows will be the first attempt to create it, and I should send an e-mail and restart Asterisk when we’re all finished.  Basically, it’s supposed to be a safety mechanism to keep from repeatedly sending the same e-mail, or restarting Asterisk once a minute if for some reason the iptables rule doesn’t “take” at first.

Again, don’t forget to make the Perl script executable, and run it manually a few times to watch the output (uncomment the commented-out “print” lines during initial testing – you can remove them once you are satisfied it’s working as it should be).  After you have run it a few times, from the Linux command prompt do iptables –list and make sure everything looks okay there.

One thing you should be aware of is that when this script detects an intrusion attempt (a connection from outside the United States), after it bans the IP address it restarts Asterisk, which will interrupt any calls in progress.  That’s deliberate; I assume you want to throw the hackers off your system right now, even if it means your users may have to re-dial their calls. Howerver, if for some reason you don’t want to do that, then you can change the line `asterisk -rx “restart now”`; to `asterisk -rx “restart when convenient”`;, which will wait until there is no usage on your system to restart Asterisk.  In that case, good luck to you if the hacker just placed a call to some $100-a-minute destination! In theory IPtables will interrupt the conversation (no audio will pass) BUT that does not mean Asterisk will tear the call down right away – when an extension “just disappears”, Asterisk tends to wait a LONG time to see if it will come back, and if it never does, well that’s what we call a “zombie” call — it just won’t die (at least not until the other end disconnects)! EDIT: If you don’t want to restart the whole system but do want to throw the hacker off NOW, see the modification by “Florent” in the comments below — I have NOT tested his changes personally, but they may do what you want.

The final step is to make this execute once a minute.  I used Webmin’s “System | Scheduled Cron Jobs” to set this up:

cron job setup using WebminBut if you are more comfortable creating a cron job from the command line, by all means, feel free to do so.

Finally, I always say that suggestions for improvement are welcome, and also, if you want to translate this into some other programming language, you have my blessing. Please be sure to test it thoroughly before relying on it, because if someone manages to hack through anyway, I’m not going to pay your phone bill! Once again, the above should be considered experimental code and is not guaranteed to do anything at all.

Mini-review of Sangoma U100 USBfxo device

 

Important
This is an edited version of a post that originally appeared on a blog called The Michigan Telephone Blog, which was written by a friend before he decided to stop blogging. It is reposted with his permission. Comments dated before the year 2013 were originally posted to his blog.

This article was originally posted in June, 2010.

I recently had the experience of trying to help someone make a Sangoma USBfxo device (model U100) work on a server that runs FreePBX and Asterisk. The advertised features of this device are as follows:

  • Dual FXO ports
  • Easy installation, no need to open up computer to install PCI/PCIe card
  • Supports up to 2 simultaneous calls
  • Compact plastic enclosure
  • Low power consumption, takes power from USB bus
  • USB 2.0 compliant (compatible with USB 1.1)

The first thing I would note is that although you don’t have to open up the computer, it’s definitely not “plug and play.” At the very least you have to install driver software, and on an Asterisk server you will also need to install and configure DADHI or ZAPTEL (unless this has already been done). Depending on your level of expertise, this might be easy, or quite daunting. I would certainly take issue with the claim of “easy installation” although I can understand how a true Linux geek might consider it a walk in the park. It wasn’t so much that there were any major hitches in the installation as that it was time consuming and required quite a bit of mental effort to figure out what needed to be done — someone who has just set up a PBX using a “load and go” distribution like Elastix, PBX in a Flash, AsteriskNOW, Trixbox, etc. might not find it all that easy to get this thing working.

The major issue we had was with the performance. We initially discovered that it was “clipping” speech severely, causing audio artifacts that are difficult to describe in print, but unpleasant to hear. We got in touch with Sangoma customer support and finally traced the problem to the built in hardware echo cancellation. By disabling the hardware echo cancellation, the speech was clear, but of course we then had mild echo. Enabling echo cancellation in Zaptel fixed that on a temporary basis, but about a week later Sangoma customer support e-mailed us and suggested that we try OSLEC, the open source echo canceler. We might have actually done that had we not discovered another issue in the meantime, that made us decide we didn’t want to mess with this unit anymore.

This new issue was that initially, it did not pick up incoming caller ID on incoming calls. We discovered that this could be fixed by changing the gain settings in Zaptel, but even when we did that it still wasn’t 100% reliable (I’d say it worked about 90% of the time). And, the downside of that was that we had to reduce the incoming gain, so that it was harder to hear callers.

We’ve used Sipura SPA-3000’s before for this same function, although they are only single line units (they have one FXS port and one FXO port) and have never had any of these issues. The main reason we tried the USBfxo was because we wanted two FXO ports, and also liked the idea that it was powered off the USB cable, and didn’t require us to have yet another device with a “wall wart” to plug in. But the difficulties with Caller ID, volume levels, and the fact that Sangoma had apparently given up on getting the hardware echo cancellation to work without distorting the audio led us to get frustrated with this device fairly quickly. The non-techies that had to make and receive calls that went through this device were not very understanding of the issues, especially since the SPA-3000’s (now superseded by the Linksys SPA-3102, which is essentially an updated version of the Sipura SPA-3000) had always worked much more reliably. We finally gave in and found another Sipura SPA-3000 on eBay and put it into service, and within a relatively short time (part of which was spent locating and installing updated firmware) it was working like a champ. Unlike the Sangoma, it detects the Caller ID 100% of the time, and we can tweak the transmit and receive gain to comfortable levels.

My personal opinion is that Sangoma should be ashamed to put their name on the USBfxo.  The hardware echo cancellation, in a word, sucks.  And one of the big reasons you’d buy a brand like Sangoma in the first place is because of the supposedly superior echo cancellation.  Echo cancellation is supposed to cancel echo, not make it sound like your words are clipped.  My guess is that the hardware echo cancellation is far too aggressive and they don’t give you any way to “tune” it — you can either enable or disable it, but that’s all.  The USBfxo is a great idea, but it needs to go back to the drawing board. Sangoma’s motto (shown on their Wiki pages, etc.) is “Because it must work!”, but apparently that motto does not imply that it must work well!

Also, a note to Sangoma customer service — next time a customer is dropping hints that they’d like you to take your defective unit back and send a replacement, you might want to be a bit more responsive to that request. We were willing to work with you up to a point but the message came through loud and clear that you really didn’t want to replace this dog of a device unless you absolutely had to.  We didn’t sign up to be beta testers, we just wanted the damn thing to work. Given Sangoma’s (perhaps undeserved) reputation we really thought you’d be more agreeable to making sure that we got a unit that worked, not making us try a bunch of different things and then ultimately told to try OSLEC, effectively giving up hope that the hardware echo cancellation would ever work properly.

Another suggestion to Sangoma (or any other manufacturer that may be listening) — most of us who did not cut our teeth on Linux would probably prefer not to have to mess with ZAPTEL or DADHI.  The nice thing about the Linksys/Sipura devices is that they sit out on the network and appear as just another SIP-based device, and in FreePBX you configure them pretty much as you would any other SIP trunk.  I’m not saying that installing any of these devices is the proverbial “piece of cake”, especially if you have never done it before, but when you have to start installing and configuring drivers, that goes outside of the realm of what I would consider easy to install. What someone really needs to come out with is an inexpensive four to six-port SIP based FXO device that sits out on your local network, like the SPA-3000/3102.

If you are in need of one or two FXO ports for your Asterisk server, my advice would be to first try one or two Sipura SPA-3000 or Linksys SPA-3102 devices (following these instructions if you are a FreePBX user) — if those do not work the way you’d like, you can always resell them on eBay and then try a more expensive solution.  If your server doesn’t have card slots (as is increasingly the case, as users turn to small computers like the Acer Aspire Revo to use as small, power-efficient PBX’s) then your choices are limited to external devices such as the aformentioned units. However, if your system can accept internal cards, then you can buy cards that provide FXO ports from several manufacturers, including Digium and Sangoma (if you need eight or more FXO ports than I believe there are other external options, but they are quite a bit more pricey and I have not really investigated them, so I won’t comment on them at this point.  However, if any manufacturer would care to send a review sample, I’d be more than happy to give it a try!). 😉

The one caveat I will add is that not every device will work on every line.  If you have a very long line from a traditional telephone company, your requirements (and experience with a particular device) may be quite different from someone who is sitting 500 feet from the central office, or someone who’s trying to take the output of a cable company’s VoIP adapter and pipe it over to the FXO card or device using twenty feet of copper wire. Just because the Sipura devices have worked better for us does not mean they will for you. I’m guessing that some people have purchased the exact same Sangoma device that we tried and were able to get it working well enough for their needs, but I just cannot recommend this device — at least not until Sangoma fixes the echo cancellation, and makes it read the incoming Caller ID reliably 100% of the time, preferably without having to change the incoming gain in DADHI or ZAPTEL.

EDIT: For more comments/opinions on this device (and on this review), see this thread on the PBX in a Flash forum.

Review of FreePBX 2.5 Powerful Telephony Solutions by Alex Robar (Packt Publishing)

 

Important
This is an edited version of a post that originally appeared on a blog called The Michigan Telephone Blog, which was written by a friend before he decided to stop blogging. It is reposted with his permission. Comments dated before the year 2013 were originally posted to his blog. In order to comply with Federal Trade Commission regulations, I am disclosing that he received a free product sample of the item under review prior to writing the review, and that any links to Amazon.com in this article are affiliate links, and if you make a purchase through one of those links I will receive a small commission on the sale.

This article was originally published in September, 2009.

Cover of FreePBX 2.5 Powerful Telephony Solutions
Cover of FreePBX 2.5 Powerful Telephony Solutions

FreePBX 2.5 Powerful Telephony Solutions by Alex Robar (Packt Publishing) explains how to set up, configure, and maintain a powerful VoIP PBX using FreePBX.  For those not familiar with FreePBX, it’s a “front end” for the Asterisk PBX software. Asterisk can be thought of as the “engine” that runs the PBX, but FreePBX is the user interface.  It basically saves you the effort of writing Asterisk configuration files and dial plans by hand.  Instead, you enter all the requisite information in FreePBX’s web-based GUI, and then when you apply the configuration changes (by clicking an orange bar at the top of the screen), FreePBX (re)writes the Asterisk dial plan and configuration files on the fly. This means that making significant changes to the call flow within the PBX, or adding new extensions or trunks, can be accomplished in a matter of seconds or minutes. It also means that you can have a fully functional PBX up and running in a few hours (perhaps even less than an hour if you’re exceptionally quick and it’s not your first installation).

As I like to do in reviews, before I begin I’ll give you a thumbnail description of what’s in each chapter (condensed from information on the Packt Publishing web site):

  • Chapter 1: Installing FreePBX – Installing FreePBX on CentOS 5.2 or Ubuntu Server 8.10
  • Chapter 2: Module Maintenance – how to install and update modules
  • Chapter 3: Devices and Extensions – explains the difference between Extensions mode and DeviceAndUser mode, and explains how to set up extensions and users. Also explains the different types of endpoints, and how to set up voicemail for a user or extension
  • Chapter 4: Trunks – discusses trunk types, setting up a new trunk, and checking trunk status
  • Chapter 5: Basic Call Targets – explains various ways to terminate calls on a FreePBX system, including Extension and Voicemail, Ring Groups, Conferences, Day Night Mode, and Phonebook Directory
  • Chapter 6: Advanced Call Targets – discusses Queues, Time conditions, and the setup of an IVR (Digital Receptionist)
  • Chapter 7: Call Routing – discusses Inbound routing, Follow Me and the VmX Locater, and Outbound routing and Least Cost Routing
  • Chapter 8: Recording Calls – focuses on everything you need to know about recording calls that pass through a FreePBX system
  • Chapter 9: Personalizing Your PBX – discusses Custom Music on Hold, Custom voice prompts, Directory search options, Customizing feature codes, Callback, Direct Inward System Access (DISA), CallerID Lookup Sources, PIN Sets, Misc applications, and Misc Destinations
  • Chapter 10: System Protection, Backup and Restoration – how to protect your system against failure, discussing both hardware methods (a good UPS and redundancy) and backups and restoration
  • Chapter 11: Security and Access Control – explains how to upgrade your operating system and Asterisk, plus various ways to secure your system against attacks

There are also four appendices:

  • Appendix A: FreePBX Modules
  • Appendix B: Feature Codes
  • Appendix C: Voicemail.conf Options
  • Appendix D: Common Trunk Configurations

I’m coming from a slightly different place in my review of this book than with other books I’ve reviewed. In this case I’ve already very familiar with the subject material, having helped set up and configure a FreePBX system that belongs to another member of my family. I was a bit afraid that because I’m already so familiar with the subject, I’d find several glaring errors or oversights. Happily, that proved not to be the case – this book is a good, solid treatment of setting up and configuring a FreePBX system. In fact, the title should have been “How to set up and maintain a FreePBX system”, because that’s exactly what this book explains.

The first thing that impressed me about this book is that there was no “filler” material. Very often, with technical books, the author really only has about 75 to 100 pages of actual material, but because publishers like to publish books that have somewhere around a couple hundred pages, the author will flesh out the book with a history of the software, a profile of the developers, comparisons with competing products, and (if they are really desperate for material) a history of the Internet. 🙂 That is not the case here. After a very short preface, the author jumps right into the subject material, starting with how to install FreePBX and then continuing through subsequent chapters with virtually everything you need to know about configuration. While not every chapter may be meaningful to every reader (personally, I’ve never had the need to record a call — so far — but it’s nice to know that FreePBX can do it), the book at least touches on all the major features of FreePBX.

If I had to make one criticism of the book, it’s that in some places it reads a little bit too much like an instruction manual. And that’s not necessarily a bad thing, because FreePBX has badly needed a good manual to assist first time users in getting it set up and running. This is the manual that should have come with FreePBX. That said, the author really doesn’t touch some of the problems frequently encountered by new users. For example, in the discussion of SIP endpoints, he notes that “SIP can be problematic when traversing firewalls and other NAT devices” and that “Configuration can be particularly troublesome if both the endpoint and the FreePBX system are behind their own firewalls” (p. 44). And there he leaves the reader hanging. There is no discussion of how to overcome the problem, nor is there a pointer to the FreePBX FAQ’s or How-To’s anywhere in the book. While many readers may not encounter this issue, a fair number will, and it would have been nice if they’d been thrown a bone, if only in the form of a pointer to the FreePBX page that addresses the issue.

In my opinion, perhaps the biggest omission is in the discussion of Trunk setup in Chapter 4. This was really the only chapter in which I got the distinct feeling that the author may have been in just a bit over his head, and perhaps did not fully grasp the subject matter covered in the chapter. Not only are there errors in his examples of dial pattern usage (p. 81 – under no circumstances would you use two pipe characters in the same dial pattern, as is shown for some of the toll-free number examples), but when discussing IAX2 and SIP trunks (p. 83), only cursory information is given about how to populate the PEER details and USER details fields. Nor is it explained that with many commercial VoIP providers, the USER context and USER details fields are not used, and should be left blank. However, in the author’s defense, I suspect that I understand why this may have happened — there probably aren’t ten people on the face of the earth that can give you a full explanation of all the options that could possibly be used in the trunk PEER and USER details fields, and when and how each option should be used. Trunk configuration is probably the most difficult part of setting up FreePBX, precisely because there’s no definitive guide on how to do it correctly. In most cases, I suspect that finding the correct options to use with any particular provider is a matter of trial and error — you make educated guesses about which options might be needed and how they should be configured, and if you find a combination that works, perhaps you post it so that others can use it. Some of the known working trunk configuration settings appear in Appendix D of the book, but there are more sample configurations available at the FreePBX web site.

I only mention this because I was hopeful that maybe someone would finally provide a really good how-to on setting up FreePBX trunks, since this is something that almost always confounds new users, and even causes experienced users to get a few (more) gray hairs on occasion. Had I been writing such a book, and had I really understood the subject, I might have given several pages to the subject of trunk configuration in general, and PEER and USER details in particular, not just a few cursory paragraphs. On the other hand, most users will probably seek out a tested and working trunk configuration for whatever provider(s) they use.  It’s not as though there isn’t any online help on the subject, but — and this is another minor criticism — for some reason the book barely mentions the availability of online help (for example, unless I missed it there is no specific mention of the FreePBX How-Tos that address several of the issues encountered by new users). This is why I say that at times the book reads like an instruction manual — it gives you all the basics, but seldom touches the “edge cases”, the little quirks and problems that may be encountered by a significant subset of users, but not by all.

However, I don’t want to leave you with the idea that this book is simply a rehash of information that could be found online — even if that were the case, it presents that information in a logical manner that is easily understandable by the reader. But, many essential functions of maintaining a FreePBX system happen outside of the FreePBX interface. For example, you cannot update your operating system or Asterisk from within the FreePBX GUI, but the book explains how to do both.  Chapters 10 and 11 (on System Protection, Backup and Restoration, and Security and Access Control) deal with functions that are at least partially handled outside of FreePBX.  In some instances the author provides useful shell scripts that automate particular tasks (for example, deleting old, outdated backups to avoid filling up the hard drive). And in many cases, the book does explain things that new users need to know, but might not know that they need to know — for example, the explanation of Codecs and the penalty involved (both in terms of system performance and call latency) in transcoding between codecs.

Anyway, the bottom line is this: Let’s say your boss wants you to set up a new office phone system using Asterisk, and gives you a couple of weeks to do it. If you have no prior experience with Asterisk, you will almost certainly want to use FreePBX (the alternative is writing dial plans and configuration files by hand, and trust me, you don’t want to do that unless you are the sort of person who enjoys writing source code for major projects, and even then you probably don’t want to do it if you’re under any sort of time deadline). And if you’re going to use FreePBX, and you don’t want to spend hours and even days ferreting out information on the Internet, you need this book. Get your boss to buy it (there’s even an e-book version if you need it right now), then just follow the instructions, chapter by chapter. In a few days time, you’ll be well on your way to becoming a FreePBX expert.  That’s partly because FreePBX is so easy to use in the first place, but also because the book tells you pretty much everything you need to know, in a very understandable manner. If you get stuck, help is available at the FreePBX web site and at other various locations on the Internet.

If you are a long-time FreePBX user, you may find that you already know much of what’s in this book, but then again it might surprise you how much can still be learned.  For example, I found several good suggestions for adding additional security to a FreePBX system in Chapter 11 of the book — and let’s face it, many of us are probably a bit lax about securing our systems to the greatest possible extent (and that could be a very costly mistake).

One other point I should make — as the title of the book implies, it deals with a particular version of FreePBX, namely version 2.5.  Of course, as so often happens with a book about software, the ink is barely dry on the paper when a new version comes out.  FreePBX 2.6 has already been offered as a release candidate, and beta versions of FreePBX 3.0 are being made available.  From a user’s standpoint, version 2.6 will be nearly identical to 2.5 – there may be a few added options and such, but for the most part they are not things that you would need to worry about, or that would detract from the accuracy of this book.  However, FreePBX 3.0 will be a major rewrite, but it’s only available in an early beta version, and unless you are an experimenter that wants to be on the bleeding edge, you don’t want it yet.  Whenever you do move to FreePBX version 3.0 — and I’d be very surprised if a full release version is much closer than a year away — much of what you’ve learned about FreePBX 2.5 and subsequent versions will still be applicable (and also, I suspect that people will be using FreePBX 2.x versions for quite some time to come).

FreePBX 2.5 Powerful Telephony Solutions by Alex Robar (Amazon affiliate link)

Mini-review of Asterisk Gateway Interface 1.4 and 1.6 Programming

 

Important
This is an edited version of a post that originally appeared on a blog called The Michigan Telephone Blog, which was written by a friend before he decided to stop blogging. It is reposted with his permission. Comments dated before the year 2013 were originally posted to his blog. In order to comply with Federal Trade Commission regulations, I am disclosing that he received a free product sample of the item under review prior to writing the review, and that any links to Amazon.com in this article are affiliate links, and if you make a purchase through one of those links I will receive a small commission on the sale.

This was originally posted in April, 2009.

I am NOT currently a PHP coder, so it’s a bit difficult for me to comment on this book. Not that the book is exclusively for PHP coders, but virtually all the examples are in PHP, so if you really want to understand what’s going on, you probably have to know at least some PHP.

So before I continue, let me run Packt Publishing’s little blub, to get the cover photo and linking out of the way:

Asterisk Gateway Interface 1.4 and 1.6 Programming Design and develop Asterisk-based VoIP telephony platforms and services using PHP and PHPAGI

  • Develop voice-enabled applications utilizing the collective power of Asterisk, PHP, and the PHPAGI class library
  • Learn basic elements of a FastAGI server utilizing PHP and PHPAGI
  • Develop new Voice 2.0 mesh-ups using the Asterisk Manager
  • Add Asterisk application development skills to your development arsenal, enriching your market offering and experience
  • Up to date for Asterisk version 1.6 and covers all previous versions

http://www.packtpub.com/asterisk-gateway-interface-programming/book

Okay, that’s the publisher’s word on the book. Now here’s my take: The one thing I thought was not right about this book was how many pages you have to go through before you get to what is supposed to be the book’s subject. The first chapter is about Installing a ‘Vanilla’ Asterisk. Now really, if someone is contemplating the purchase of this book, don’t you think it’s likely that they have Asterisk (in some form) already installed?

The second chapter is on Basic IVR Development: Using the Asterisk DialPlan and while I suppose this chapter is somewhat useful in laying the groundwork for what is to come, by the time you are through with it you are already up to page 53. And then comes the next chapter: More IVR Development: Input, Recordings, and Call Control. And that chapter ends on page 73. This is a problem because the last regular page of the book, at the end of Chapter 10, is page 191 – which basically means that nearly half the book is devoted to installing Asterisk from scratch, and then creating an IVR the old-fashioned way (if I want an IVR I’ll create it in FreePBX, thank you very much). And there is virtually nothing about AGI programming in those initial three chapters.

Finally, in Chapter 4, starting on page 75, we get into A Primer to AGI: Asterisk Gateway Interface – and this is where the book hits its stride and keeps on going. If you know PHP but know next to nothing about Asterisk AGI programming, this book will teach you the basics of what you need to know to start writing AGI code – but perhaps just as important, it will teach you good coding practices. It explains when you should try to put code in the Asterisk DialPlan as opposed to putting it in an AGI script. It explains why certain languages are much better than others when you are writing an AGI script. And it will teach you the fundamental rationale behind AGI scripts – if you are a seldom-coder like me, you probably think that calling an AGI script is simply calling a bit of code written in some other programming language. Nothing could be further from the truth.

AGI scripts are actually quite a bit more powerful than many Asterisk users realize – I think that theoretically you could do most of your Asterisk programming from inside an AGI script, but that doesn’t mean you should. One of the things that the AGI shines at is obtaining data that’s not normally available to Asterisk – for example, information from a local or remote database, or even from a FTP server or web-based service. You’ve seen web-based “click to call” services like Jajah? That’s basically obtaining data from a web server so that Asterisk can initiate a callback, and this book shows you how something like that is set up.

I’ve heard it said that in many books there is one chapter that epitomizes the entire volume. In this case, there are about five chapters that are the heart of the book, but you will learn a great deal in those five chapters. Even without knowing a lick of PHP, I came away with a greater understanding of what AGI scripts are, and when and how they should be used. Had I actually been able to understand the PHP examples, I think I’d have had a much greater appreciation for the book. The author definitely knows his subject; and perhaps that’s why once he gets into it you feel like you’re getting solid information.

So I guess I would say this: If you judge the value of a book solely on page count, and you don’t need to know how to install Asterisk from scratch or set up an IVR, then you may feel a bit disappointed by this book (if that’s really the case, do yourself a favor and skip the first three chapters). But if you judge a text on how quickly it can bring you up to speed on a subject, and you want to learn Asterisk AGI programming, I think you’ll like it. Since the book relies so heavily on PHP, I think the first three chapters might have been better spent on an introduction to PHP – if the author would have covered that as well as he covers the Asterisk Gateway Interface, I’m sure I would have had a greater appreciation of this volume.

One other thing I noted is that this book seems to be very current – It’s not talking about Asterisk as it was two years ago. It was first published in January, 2009. That’s important when you are looking for information related to Asterisk, which can change quite a bit in a couple of years.

Asterisk Gateway Interface 1.4 and 1.6 Programming (Amazon affiliate link) by Nir Simionovich
Chapter List:

Table of Contents
Preface
Chapter 1: Installing a ‘Vanilla’ Asterisk
Chapter 2: Basic IVR Development: Using the Asterisk DialPlan
Chapter 3: More IVR Development: Input, Recordings, and Call Control
Chapter 4: A Primer to AGI: Asterisk Gateway Interface
Chapter 5: AGI Scripting with PHP
Chapter 6: PHPAGI: An AGI Class Library in PHP
Chapter 7: FastAGI: AGI as a TCP Server
Chapter 8: AMI: The Asterisk Manager Interface
Chapter 9: Final Programming Project
Chapter 10: Scaling Asterisk Applications
Index

Or if you’d like to see a complete Table of Contents with subheadings, in outline form, go here.

Author Nir Simionovich’s blog

Linksys and Sipura adapter users – check your RTP Packet Size and Network Jitter Level

 

Important
This is an edited version of a post that originally appeared on a blog called The Michigan Telephone Blog, which was written by a friend before he decided to stop blogging. It is reposted with his permission. Comments dated before the year 2013 were originally posted to his blog.

Edit: Reader Christopher Woods notes in a comment that the following is also applicable to at least some models of Linksys phones, e.g. SPA942 and SPA962.

Do you use a Linksys or Sipura VoIP adapter? Do the people you are talking to ever complain about your voice breaking up, or missing or dropped syllables, or unexplained clicks or noise?

There is an obscure setting in Linksys/Sipura VoIP adapters that is usually set incorrectly for most applications, at least on a factory-fresh adapter. Go to the SIP tab and check the RTP Packet Size – for most users, it should be set to 0.020 rather than the factory preset of 0.030. If you are running a connection where latency is critical (say you have a cable or satellite box that requires a phone connection to “phone home”, or you are trying to use a FAX machine) then you may even wish to set this to 0.010, which further reduces latency, at the expense of using a bit more bandwidth. In any case, the default 0.030 is not the correct setting when using the most commonly-used codecs. For more discussion of this issue, see this thread at DSLreports.com, which discusses how the RTP Packet Size and Network Jitter Level settings can be tweaked to achieve lower latency, along with the tradeoffs.

Be aware that the RTP Packet Size setting is found under the SIP tab, and that setting is applied to all lines served through that adapter. However, the Network Jitter Level can be set individually for each line, under the Line tabs. One interesting comment in the above-mentioned thread is that if a provider forces you to use a low-bandwidth codec, decreasing the RTP Packet Size may increase the quality of your calls, but again at the expense of increasing bandwidth used.

Changing the RTP Packet Size on one VoIP adapter resolved a few strange issues with audio quality. In this case the adapter was being used to connect to an Asterisk box on the same local network, so bandwidth usage wasn’t an issue. We set the RTP Packet Size to 0.020 and the Network Jitter Level to low, and it made a noticeable difference in the reduction of strange noises and breakups heard by the party on the other end of the conversation. However, changing the Network Jitter Level isn’t as critical as changing the RTP Packet Size, and in fact, changing the Network Jitter Level may be entirely the wrong thing to do on certain types of connections (probably not a good idea if your adapter is connected through a Wireless ISP, for example).

I must thank Paul Timmins for being the first to point out that the Linksys PAP2 has a default packet size of 0.030, which is incompatible with the uLaw (G711u) codec (or at least in violation of the standard). With that lead, I then discovered other articles (including the discussion thread linked above) that said essentially the same thing. So check those adapter settings, folks!

(And by the way, this advice probably does apply to some other makes of VoIP adapters, and even some IP telephones, but since I don’t have any readily available to look at, I can’t say for sure. If you know of any others that need to have a similar setting tweaked, please feel free to add a comment to this post).

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