Category: Trixbox

Why Asterisk-based GUI software should make better use of regular expressions

 

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.

Having recently discovered that Asterisk supports regular expressions (with some limitations), I got to wondering why the FreePBX-based GUI’s don’t make better use of them.  For example, in FreePBX, Trixbox, Elastix, etc., the Outbound Route pages all support entering patterns that are then directly used in Asterisk dialplans, in a most inefficient manner.  Let me give you an example.  Let’s say you have created an Outbound Route for toll-free calls, and in the “Dial Patterns that will use this Route” section you have entered these patterns:

1800NXXXXXX
1844NXXXXXX
1855NXXXXXX
1866NXXXXXX
1877NXXXXXX
1888NXXXXXX
800NXXXXXX
844NXXXXXX
855NXXXXXX
866NXXXXXX
877NXXXXXX
888NXXXXXX

Let’s further say that you have specified three trunks in the “Trunk Sequence for Matched Routes” section (this is actually conservative given that there are a number of ways you can route toll-free calls for free).  Anyway, this is an actual representative context that would be generated in extensions_additional.conf:

[outrt-5] ; TollFree
include => outrt-5-custom
exten => _1800NXXXXXX,1,Macro(user-callerid,SKIPTTL,)
exten => _1800NXXXXXX,n,Noop(Calling Out Route: TollFree)
exten => _1800NXXXXXX,n,Set(MOHCLASS=${IF($[“${MOHCLASS}”=””]?default:${MOHCLASS})})
exten => _1800NXXXXXX,n,Set(_NODEST=)
exten => _1800NXXXXXX,n,Macro(record-enable,${AMPUSER},OUT,)
exten => _1800NXXXXXX,n,Macro(dialout-trunk,7,${EXTEN},)
exten => _1800NXXXXXX,n,Macro(dialout-trunk,10,${EXTEN},)
exten => _1800NXXXXXX,n,Macro(dialout-trunk,8,${EXTEN},)
exten => _1800NXXXXXX,n,Macro(outisbusy,)
exten => _1844NXXXXXX,1,Macro(user-callerid,SKIPTTL,)
exten => _1844NXXXXXX,n,Noop(Calling Out Route: TollFree)
exten => _1844NXXXXXX,n,Set(MOHCLASS=${IF($[“${MOHCLASS}”=””]?default:${MOHCLASS})})
exten => _1844NXXXXXX,n,Set(_NODEST=)
exten => _1844NXXXXXX,n,Macro(record-enable,${AMPUSER},OUT,)
exten => _1844NXXXXXX,n,Macro(dialout-trunk,7,${EXTEN},)
exten => _1844NXXXXXX,n,Macro(dialout-trunk,10,${EXTEN},)
exten => _1844NXXXXXX,n,Macro(dialout-trunk,8,${EXTEN},)
exten => _1844NXXXXXX,n,Macro(outisbusy,)
exten => _1855NXXXXXX,1,Macro(user-callerid,SKIPTTL,)
exten => _1855NXXXXXX,n,Noop(Calling Out Route: TollFree)
exten => _1855NXXXXXX,n,Set(MOHCLASS=${IF($[“${MOHCLASS}”=””]?default:${MOHCLASS})})
exten => _1855NXXXXXX,n,Set(_NODEST=)
exten => _1855NXXXXXX,n,Macro(record-enable,${AMPUSER},OUT,)
exten => _1855NXXXXXX,n,Macro(dialout-trunk,7,${EXTEN},)
exten => _1855NXXXXXX,n,Macro(dialout-trunk,10,${EXTEN},)
exten => _1855NXXXXXX,n,Macro(dialout-trunk,8,${EXTEN},)
exten => _1855NXXXXXX,n,Macro(outisbusy,)
exten => _1866NXXXXXX,1,Macro(user-callerid,SKIPTTL,)
exten => _1866NXXXXXX,n,Noop(Calling Out Route: TollFree)
exten => _1866NXXXXXX,n,Set(MOHCLASS=${IF($[“${MOHCLASS}”=””]?default:${MOHCLASS})})
exten => _1866NXXXXXX,n,Set(_NODEST=)
exten => _1866NXXXXXX,n,Macro(record-enable,${AMPUSER},OUT,)
exten => _1866NXXXXXX,n,Macro(dialout-trunk,7,${EXTEN},)
exten => _1866NXXXXXX,n,Macro(dialout-trunk,10,${EXTEN},)
exten => _1866NXXXXXX,n,Macro(dialout-trunk,8,${EXTEN},)
exten => _1866NXXXXXX,n,Macro(outisbusy,)
exten => _1877NXXXXXX,1,Macro(user-callerid,SKIPTTL,)
exten => _1877NXXXXXX,n,Noop(Calling Out Route: TollFree)
exten => _1877NXXXXXX,n,Set(MOHCLASS=${IF($[“${MOHCLASS}”=””]?default:${MOHCLASS})})
exten => _1877NXXXXXX,n,Set(_NODEST=)
exten => _1877NXXXXXX,n,Macro(record-enable,${AMPUSER},OUT,)
exten => _1877NXXXXXX,n,Macro(dialout-trunk,7,${EXTEN},)
exten => _1877NXXXXXX,n,Macro(dialout-trunk,10,${EXTEN},)
exten => _1877NXXXXXX,n,Macro(dialout-trunk,8,${EXTEN},)
exten => _1877NXXXXXX,n,Macro(outisbusy,)
exten => _1888NXXXXXX,1,Macro(user-callerid,SKIPTTL,)
exten => _1888NXXXXXX,n,Noop(Calling Out Route: TollFree)
exten => _1888NXXXXXX,n,Set(MOHCLASS=${IF($[“${MOHCLASS}”=””]?default:${MOHCLASS})})
exten => _1888NXXXXXX,n,Set(_NODEST=)
exten => _1888NXXXXXX,n,Macro(record-enable,${AMPUSER},OUT,)
exten => _1888NXXXXXX,n,Macro(dialout-trunk,7,${EXTEN},)
exten => _1888NXXXXXX,n,Macro(dialout-trunk,10,${EXTEN},)
exten => _1888NXXXXXX,n,Macro(dialout-trunk,8,${EXTEN},)
exten => _1888NXXXXXX,n,Macro(outisbusy,)
exten => _800NXXXXXX,1,Macro(user-callerid,SKIPTTL,)
exten => _800NXXXXXX,n,Noop(Calling Out Route: TollFree)
exten => _800NXXXXXX,n,Set(MOHCLASS=${IF($[“${MOHCLASS}”=””]?default:${MOHCLASS})})
exten => _800NXXXXXX,n,Set(_NODEST=)
exten => _800NXXXXXX,n,Macro(record-enable,${AMPUSER},OUT,)
exten => _800NXXXXXX,n,Macro(dialout-trunk,7,${EXTEN},)
exten => _800NXXXXXX,n,Macro(dialout-trunk,10,${EXTEN},)
exten => _800NXXXXXX,n,Macro(dialout-trunk,8,${EXTEN},)
exten => _800NXXXXXX,n,Macro(outisbusy,)
exten => _844NXXXXXX,1,Macro(user-callerid,SKIPTTL,)
exten => _844NXXXXXX,n,Noop(Calling Out Route: TollFree)
exten => _844NXXXXXX,n,Set(MOHCLASS=${IF($[“${MOHCLASS}”=””]?default:${MOHCLASS})})
exten => _844NXXXXXX,n,Set(_NODEST=)
exten => _844NXXXXXX,n,Macro(record-enable,${AMPUSER},OUT,)
exten => _844NXXXXXX,n,Macro(dialout-trunk,7,${EXTEN},)
exten => _844NXXXXXX,n,Macro(dialout-trunk,10,${EXTEN},)
exten => _844NXXXXXX,n,Macro(dialout-trunk,8,${EXTEN},)
exten => _844NXXXXXX,n,Macro(outisbusy,)
exten => _855NXXXXXX,1,Macro(user-callerid,SKIPTTL,)
exten => _855NXXXXXX,n,Noop(Calling Out Route: TollFree)
exten => _855NXXXXXX,n,Set(MOHCLASS=${IF($[“${MOHCLASS}”=””]?default:${MOHCLASS})})
exten => _855NXXXXXX,n,Set(_NODEST=)
exten => _855NXXXXXX,n,Macro(record-enable,${AMPUSER},OUT,)
exten => _855NXXXXXX,n,Macro(dialout-trunk,7,${EXTEN},)
exten => _855NXXXXXX,n,Macro(dialout-trunk,10,${EXTEN},)
exten => _855NXXXXXX,n,Macro(dialout-trunk,8,${EXTEN},)
exten => _855NXXXXXX,n,Macro(outisbusy,)
exten => _866NXXXXXX,1,Macro(user-callerid,SKIPTTL,)
exten => _866NXXXXXX,n,Noop(Calling Out Route: TollFree)
exten => _866NXXXXXX,n,Set(MOHCLASS=${IF($[“${MOHCLASS}”=””]?default:${MOHCLASS})})
exten => _866NXXXXXX,n,Set(_NODEST=)
exten => _866NXXXXXX,n,Macro(record-enable,${AMPUSER},OUT,)
exten => _866NXXXXXX,n,Macro(dialout-trunk,7,${EXTEN},)
exten => _866NXXXXXX,n,Macro(dialout-trunk,10,${EXTEN},)
exten => _866NXXXXXX,n,Macro(dialout-trunk,8,${EXTEN},)
exten => _866NXXXXXX,n,Macro(outisbusy,)
exten => _877NXXXXXX,1,Macro(user-callerid,SKIPTTL,)
exten => _877NXXXXXX,n,Noop(Calling Out Route: TollFree)
exten => _877NXXXXXX,n,Set(MOHCLASS=${IF($[“${MOHCLASS}”=””]?default:${MOHCLASS})})
exten => _877NXXXXXX,n,Set(_NODEST=)
exten => _877NXXXXXX,n,Macro(record-enable,${AMPUSER},OUT,)
exten => _877NXXXXXX,n,Macro(dialout-trunk,7,${EXTEN},)
exten => _877NXXXXXX,n,Macro(dialout-trunk,10,${EXTEN},)
exten => _877NXXXXXX,n,Macro(dialout-trunk,8,${EXTEN},)
exten => _877NXXXXXX,n,Macro(outisbusy,)
exten => _888NXXXXXX,1,Macro(user-callerid,SKIPTTL,)
exten => _888NXXXXXX,n,Noop(Calling Out Route: TollFree)
exten => _888NXXXXXX,n,Set(MOHCLASS=${IF($[“${MOHCLASS}”=””]?default:${MOHCLASS})})
exten => _888NXXXXXX,n,Set(_NODEST=)
exten => _888NXXXXXX,n,Macro(record-enable,${AMPUSER},OUT,)
exten => _888NXXXXXX,n,Macro(dialout-trunk,7,${EXTEN},)
exten => _888NXXXXXX,n,Macro(dialout-trunk,10,${EXTEN},)
exten => _888NXXXXXX,n,Macro(dialout-trunk,8,${EXTEN},)
exten => _888NXXXXXX,n,Macro(outisbusy,)

; end of [outrt-5]

That is 112 lines of dialplan generated by only 12 lines of pattern in the Outbound Route and three trunk selections.  If you have more patterns and/or more trunks, the generated dialplan can grow MUCH larger.  At one point I had a dialplan so large that whenever I did an “orange bar” reload, it took two to three minutes to finish reloading.  Clearly, there is something wrong here.

Now, let’s suppose that instead of, or better yet, in addition to the current Outbound Route pages, we had pages called “Outbound Route – Regular Expression”.  It would eliminate the “Dial Patterns that will use this Route” section, replacing it with a “Regular Expression to select this Route” text box, and it would also have a text box that says “Extensions allowed to use this route (separated by the | character – RegEx patterns may be used).

Let’s say that the contents of the first text box are ^1?8(00|44|55|66|77|88)[2-9][0-9]{6}$ and are stored in variable string1. And let’s say that the contents of the second text box are 10[0-4]X|106[5-8] (specifying that extensions 1000-1049 and 1065-1068 may use this route) and are stored in string 2. Assuming that the part of the dialplan that processes this were in some kind of macro, and the called number is in the variable OUTNUM, you could perhaps do something like this:

exten => s,n,ExecIf($[${REGEX(“${string2}” ${CALLERID(num)})} != 1]?MacroExit())
exten => s,n,ExecIf($[${REGEX(“${string1}” ${OUTNUM})} != 1]?MacroExit())
exten => s,n,Macro(dialout-trunk,7,${OUTNUM},)
exten => s,n,Macro(dialout-trunk,10,${OUTNUM},)
exten => s,n,Macro(dialout-trunk,8,${OUTNUM},)
exten => s,n,Macro(outisbusy,)

The first line bails out of the macro if the Caller ID doesn’t match one of the permitted extensions (this would assume you by default use a regex that matches any extension on your system, for example ^[0-9]{4}$ would match any 4 digit extension number. If you don’t assume this, then you’d need an extra bit of code to skip a null value in string2).

The second line bails if the called number doesn’t match the pattern in string1.

If the called number does match on the regular expression for this route, and the call is from a permitted extension, then the three trunks are tried. Note that now you only need to list each of the three trunks once.

Granted this doesn’t account for every line in the original context, but the other lines could also be transferred into the macro, with a bit of thought. Wouldn’t this be simpler for those who know how to use regular expressions? And, it has the potential to generate MUCH shorter dialplans.

Those who understand the power of regular expressions and who have long or complicated dialplans would appreciate the ability to use regular expressions!  If anyone implements this, don’t be stingy with the maximum string length in the Regular Expression text boxes; I guarantee you that 256 characters is not enough, and neither is 1,024 characters! At a MINIMUM I would say that you need to allow a 4,096 character (4K) regular expression, and while I personally can’t conceive of anyone exceeding that, my bet is that sooner or later someone will.  Remember, you are potentially saving a whole bunch of memory and drive space by allowing regular expressions, so you don’t have to be miserly on the space allotted for the expressions themselves.

I no longer recommend using Asterisk’s Google Voice support — try these methods instead!

 

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 (May, 2018): FreePBX and Asterisk users that wish to continue using Google Voice after Google drops XMPP support should go here: How to use Google Voice with FreePBX and Asterisk without using XMPP or buying new hardware.

This article was originally written in January of 2012, and has been heavily edited in an attempt to update it a bit.

Not that anyone probably cares what I think, but anyone who regularly reads this blog (or any of the other VoIP-related blog that cover Asterisk) may have noticed that prior to the release of Asterisk 11, Asterisk’s support for Google Voice had become less and less reliable over time, particularly for incoming calls. You have to do all sorts of “tricks” to make it work, and these usually involve adding delays that don’t always fix the problem, inconvenience your callers, and possibly cause more hangups as people get tired of waiting for you to answer the phone.

Therefore, I suggest that if you are using a version of Asterisk earlier than Asterisk 11, you stop using Asterisk’s Google Voice support completely. Assuming that you feel you must keep using an older version of Asterisk, I suggest trying one or more of the following:

  1. Use YATE as a gateway between Asterisk and Google Voice. See Using YATE to overcome Google Voice issues in FreeSWITCH and Asterisk, this article and this forum thread on YATE in a Flash, and this thread on YATE Tips & Tricks). YATE is what powers Bill Simon’s gateway (mentioned below). See comments by Bill and pianoquintet under this article.
  2. Use Bill Simon’s Google Voice-SIP gateway to handle your Google Voice calls. Some people may not want to rely on an external service for this, while others may very much appreciate having the option. I mention it for those in the latter group. For more information see Bill Simon’s Free SIP-to-XMPP Gateway Easily Puts Google Voice on Your VoIP Phone (Voxilla). While the linked articles talk about using the gateway with a SIP device, it can be used as an Asterisk trunk.  EDIT: As of April 7, 2015 the Google Voice Gateway has been relaunched and there is now a one-time fee to sign up.
  3. If your only issue is with incoming calls, you could use a DID to bring the calls into your system.  But keep in mind that Google Voice does not like it when calls are answered the moment they connect, so in your FreePBX Inbound Route be sure to set the “Pause Before Answer” option to at least 1.  I have found that a 1 second pause is sufficient, but I’m not saying that is the correct value for everyone, or even that everyone will need to include such a pause (some DID providers may delay the call sufficiently before connecting through to your system that the pause isn’t needed).

At this point, any of those would likely produce better results than using the Google Voice support in any version of Asterisk prior to Asterisk 11.

EVERYTHING in this article is my personal opinion.  Nothing here should be taken as a statement of fact.

EDIT:  Ward Mundy reports that he just may have found a workaround for the incoming calls issue — see this thread in the PBX in a Flash forum.

How to install the BIND DNS Server using Webmin, so Asterisk extensions (hopefully) will work even when your Internet connection fails

 

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.

If you run Asterisk you may have encountered this issue: Your Internet connection goes down, and so does your ability to call from extension to extension, even between extensions on your local network. This is a long-standing bug in Asterisk (exactly the sort of bug that drives people to try alternatives such as FreeSWITCH) but let’s say that for whatever reason you need to stick with Asterisk, so you’d like to find a way to make that bug go away.  Without going into all the technical details, the reason that calls fail is that Asterisk can’t access a DNS server.  I’ve read several reports that say the easiest solution is to install the BIND DNS server  on the same machine as your Asterisk server.  If you are also running Webmin on the server, installing and configuring BIND is a relative piece of cake.  So here’s how it’s done.  Please note that most of the images below can be enlarged by clicking on them, and that I have installed the StressFree theme in Webmin, so if it looks a little different from what you’re used to seeing, that’s probably why.

To start with, log into Webmin, click on “Servers”, then click on “BIND DNS Server” (if you don’t find it there, try looking in “Un-used Modules”):

Webmin Servers page — click on "BIND DNS Server"

Assuming you have not previously installed BIND, you’ll get a screen like this.  Just click where it says “Click here”:

Webmin BIND DNS Server error page — click where it says "Click here"

You will then see this screen come up as BIND is installed. Just let it run to completion and (assuming it installs successfully) click on “Return to BIND DNS Server” at the bottom of the page:

Webmin "Install Package" page — click on "Return to BIND DNS Server"

Next, because you don’t yet have an /etc/named.conf file, you’ll see this page.  Click the button for “Setup nameserver for internal non-internet use only” (don’t worry, we’ll fix it in the next steps), then click the bar that says “Create Primary Configuration File and Start Nameserver”:

Webmin — click "Setup nameserver for internal non-internet use only"

At this point BIND is installed and running, but it probably isn’t doing what you want it to, and your system isn’t using it. So the first thing we need to do is tell it where to go when it needs to do a DNS lookup. You should be seeing a page that looks like this — click on “Forwarding and Transfers”:

Webmin BIND DNS Server page — click on "Forwarding and Transfers"

When you get to the following screen, check “Yes” next to “Lookup directly if forwarders cannot?”  You also need to enter one or more addresses of DNS servers that BIND can access when it needs to pull a DNS record.  You might want to give some thought to which DNS servers you want to use, and in what order, before you start entering them. You can enter up to three IP addresses of DNS servers, and then click “Save”. This will throw you out to the previous screen, and if by some chance you want to enter even more DNS servers, you can click on “Forwarding and Transfers” again to come back and enter up to three more servers, until you are finished.  In this example, I have already entered the IP addresses of my router’s DNS Server as the top priority pick,  followed by two Google DNS Server addresses.

Webmin — BIND DNS Server — Forwarding and Transfers page

Once you have done this, you are through configuring BIND directly, but there are two more things we need to do. The first is to make sure that the BIND server starts each time we restart the machine. To do that, go to Webmin’s “System” page and then click on “Bootup and Shutdown”:

Webmin System page — click on "Bootup and Shutdown"

This is a long page so I’m not showing all of it — what you have to do is find the entry for named and check the box next to it:

Webmin Bootup and Shutdown page — check the box next to "named"

Then go to the bottom of the page and click “Start on Boot”:

Bottom of Webmin Bootup and Shutdown page — click "Start on Boot"

At this point BIND is running, and should be using the correct DNS servers, and is set to start at bootup, but your server still isn’t using it for its DNS queries. To get it to do that, go to Webmin’s “Networking” page and click on “Network Configuration”:

Webmin Networking page — click on "Network Configuration"

Once on the Network Configuration page, click on “Hostname and DNS Client”:

Webmin Network Configuration page — click on "Hostname and DNS Client"

Once on the Hostname and DNS Client page, what you need to do is make the first entry in the DNS Servers list 127.0.0.1. If you trust BIND to always be operating, that’s the only entry you need. I didn’t quite trust BIND that much (actually, what I didn’t trust was my ability to set this up correctly) so I set the DNS server in the router as the secondary DNS address. You could use any DNS server as the secondary, or you could choose to just enter the 127.0.0.1 address to use BIND and let it go at that. Personally, I feel a lot more comfortable having a “fallback” DNS. Don’t forget to click “Save” when you are finished making changes here:

Webmin Hostname and DNS Client page - 127.0.0.1 must be first DNS server

That’s all there is to it, as far as I know (if you think I’ve missed anything or done something wrong, the comment section is open!). If you’re like me, the next question you will have is, “How do I know it’s working?” And the easiest way to do that is to go to a Linux command prompt and “dig” some site you have not been to recently twice in a row. Here’s an example, using cnn.com — the part we are interested in is in red:

dig cnn.com

; <<>> DiG 9.3.6-P1-RedHat-9.3.6-4.P1.el5_5.3 <<>> cnn.com
;; global options:  printcmd
;; Got answer:
;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 8274
;; flags: qr rd ra; QUERY: 1, ANSWER: 6, AUTHORITY: 13, ADDITIONAL: 9

;; QUESTION SECTION:
;cnn.com.                       IN      A

;; ANSWER SECTION:
cnn.com.                287     IN      A       157.166.224.25
cnn.com.                287     IN      A       157.166.224.26
cnn.com.                287     IN      A       157.166.226.25
cnn.com.                287     IN      A       157.166.226.26
cnn.com.                287     IN      A       157.166.255.18
cnn.com.                287     IN      A       157.166.255.19

;; AUTHORITY SECTION:
.                       76691   IN      NS      i.root-servers.net.
.                       76691   IN      NS      j.root-servers.net.
.                       76691   IN      NS      k.root-servers.net.
.                       76691   IN      NS      l.root-servers.net.
.                       76691   IN      NS      m.root-servers.net.
.                       76691   IN      NS      a.root-servers.net.
.                       76691   IN      NS      b.root-servers.net.
.                       76691   IN      NS      c.root-servers.net.
.                       76691   IN      NS      d.root-servers.net.
.                       76691   IN      NS      e.root-servers.net.
.                       76691   IN      NS      f.root-servers.net.
.                       76691   IN      NS      g.root-servers.net.
.                       76691   IN      NS      h.root-servers.net.

;; ADDITIONAL SECTION:
b.root-servers.net.     386178  IN      A       192.228.79.201
d.root-servers.net.     402826  IN      A       128.8.10.90
d.root-servers.net.     230000  IN      AAAA    2001:500:2d::d
f.root-servers.net.     370827  IN      A       192.5.5.241
g.root-servers.net.     463754  IN      A       192.112.36.4
h.root-servers.net.     374116  IN      A       128.63.2.53
h.root-servers.net.     517382  IN      AAAA    2001:500:1::803f:235
j.root-servers.net.     185528  IN      A       192.58.128.30
j.root-servers.net.     578747  IN      AAAA    2001:503:c27::2:30

;; Query time: 26 msec
;; SERVER: 127.0.0.1#53(127.0.0.1)
;; WHEN: Fri Sep 16 12:45:41 2011
;; MSG SIZE  rcvd: 512

# dig cnn.com

; <<>> DiG 9.3.6-P1-RedHat-9.3.6-4.P1.el5_5.3 <<>> cnn.com
;; global options:  printcmd
;; Got answer:
;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 8277
;; flags: qr rd ra; QUERY: 1, ANSWER: 6, AUTHORITY: 13, ADDITIONAL: 9

;; QUESTION SECTION:
;cnn.com.                       IN      A

;; ANSWER SECTION:
cnn.com.                223     IN      A       157.166.255.19
cnn.com.                223     IN      A       157.166.224.25
cnn.com.                223     IN      A       157.166.224.26
cnn.com.                223     IN      A       157.166.226.25
cnn.com.                223     IN      A       157.166.226.26
cnn.com.                223     IN      A       157.166.255.18

;; AUTHORITY SECTION:
.                       76627   IN      NS      c.root-servers.net.
.                       76627   IN      NS      d.root-servers.net.
.                       76627   IN      NS      e.root-servers.net.
.                       76627   IN      NS      f.root-servers.net.
.                       76627   IN      NS      g.root-servers.net.
.                       76627   IN      NS      h.root-servers.net.
.                       76627   IN      NS      i.root-servers.net.
.                       76627   IN      NS      j.root-servers.net.
.                       76627   IN      NS      k.root-servers.net.
.                       76627   IN      NS      l.root-servers.net.
.                       76627   IN      NS      m.root-servers.net.
.                       76627   IN      NS      a.root-servers.net.
.                       76627   IN      NS      b.root-servers.net.

;; ADDITIONAL SECTION:
b.root-servers.net.     386114  IN      A       192.228.79.201
d.root-servers.net.     402762  IN      A       128.8.10.90
d.root-servers.net.     229936  IN      AAAA    2001:500:2d::d
f.root-servers.net.     370763  IN      A       192.5.5.241
g.root-servers.net.     463690  IN      A       192.112.36.4
h.root-servers.net.     374052  IN      A       128.63.2.53
h.root-servers.net.     517318  IN      AAAA    2001:500:1::803f:235
j.root-servers.net.     185464  IN      A       192.58.128.30
j.root-servers.net.     578683  IN      AAAA    2001:503:c27::2:30

;; Query time: 1 msec
;; SERVER: 127.0.0.1#53(127.0.0.1)
;; WHEN: Fri Sep 16 12:46:45 2011
;; MSG SIZE  rcvd: 512

Notice how on the first run, it takes 26 msec to do the lookup, because BIND doesn’t have that address cached yet, whereas on the second run it only takes 1 msec to do the lookup!  Could that perhaps improve system performance?  I’ll bet it could! And the SERVER line tells us that it is indeed using our BIND server (127.0.0.1) – if it were using, say, our router’s DNS server then that line would show this:

;; SERVER: 192.168.0.1#53(192.168.0.1)

The idea here is that when your Internet connection takes a dive, Asterisk will still be finding a working DNS server and therefore won’t tank.  That, at least, is the theory I’ve seen on several web sites.  The ONLY thing I am showing here is how to set up BIND using Webmin, and I won’t even guarantee that I’m doing that 100% correctly.  I definitely do not guarantee that it will actually work as intended — you’ll have to test that yourself.  Doing a real test would mean disconnecting your cable or DSL modem, etc. from your router for several hours or days to see if the phones continue to work, and in most households or businesses that idea will go over like a lead balloon.  However, feel free to give it a good test if you like and report the results in the comments.

You may wonder why I selected “Setup nameserver for internal non-internet use only” in the fourth screenshot.  Obviously, that description is not entirely accurate.  The real difference is that if you select that instead of the default “Setup as an internet name server, and download root server information”, it won’t create a “root” DNS zone, which you simply don’t need for this application.  You can use the other option if you want to, but it will download additional information and increase the complexity of your setup.  Either way, you should be able to access the Internet, because we set up DNS forwarding.  If by some chance this BIND server is going to act as a nameserver for your entire network, and you don’t mind the additional traffic and complexity (and it’s the additional traffic that scares me the most, since I have no idea what it’s actually downloading nor how often it’s doing it), then by all means feel free to use the second option.  All I will say is that I used the first. and it works fine, and I’ve seen at least one instance where this same thing is set up using a method other than Webmin, and except for the order of statements it uses an /etc/named.conf file that is identical to what Webmin produces when configured as I have shown here (in other words, no “zones” at all).  I’m just waiting for some Linux purist to say this isn’t the “right” way to do this but keep the goal in mind here — all we are trying to do is work around a bug in Asterisk that should have been fixed years ago, not set up a DNS server to feed an entire subnet.  But again, you can feel free to use whichever of the options you like — it should work either way.

(By the way, if after reading the above you have “setup remorse” — you know, that feeling you get after you’ve installed something that you should have picked a different option — you can get a “do-over” by simply deleting or moving/renaming /etc/named.conf.  If you then exit Webmin’s BIND module and come back in, it should see that named.conf doesn’t exist and start you over at the fourth screen shown above.  Of course, you will lose anything you have already configured from within that module.  If you originally selected the option to download the root server information, I think that’s at least partly stored in the file /etc/db.cache, so you could move or remove that file to make sure it’s not used, however I’m not sure if any other files are or were also downloaded.  That particular file is very small so I’m not worried about that one per se, it’s just that the way things are worded on a couple of pages I read, I don’t know if that’s all it downloads, or if at some point in the middle of the night it rises up and tried to cache all the DNS information for the Internet, or just exactly what it does.  Sometimes I wish people would just give a sentence or two of additional information, so you have a better idea of what’s the right thing to do when you’re setting up something like this.)

Now, if you are a True Linux Geek who somehow stumbled across this article, and are disappointed that it isn’t much more complicated, I’ll refer you to this page.  If you can figure all THAT out, you should be getting paid the big bucks as the networking expert that you are! 🙂

How to show the source DID in FreePBX call detail reports

 

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 July, 2011. This functionality may or may not be already present in newer versions of FreePBX.

NOTE:  If you have installed a separate FreePBX CDR report tool module and it does not already include this functionality, you may need to do this before you follow these instructions: Copy this article (starting with the paragraph just before “[custom-from-trunk-accountcode]”) and paste it into a text editor, then use the editor’s search-and-replace to search for the string “accountcode” and change every occurrence of it to “did” (without the quotes in both cases), and follow those instructions.  That way you’ll be saving your DID information to the did field created in the CDR database by the module. I cannot tell you if you need to do this or not, because the last version of FreePBX I used was 2.9, and I have no plans to upgrade.

Another thing I am posting just so I can find it later…

Many people have wished for a way to see the source DID in FreePBX call detail reports.  It turns out there is a pretty easy way to do it IF you are not using Account Codes in your system, and have no plans to do so.  Many FreePBX users never use account codes so the following method will work for them.

Asterisk stores its CDR data in /var/log/asterisk/cdr-csv/Master.csv.  If you know the format in which it stores its data, you realize that the first field, accountcode, isn’t used on most systems.  Since Asterisk currently doesn’t store the source DID for a call, we can repurpose the accountcode field to store our source DID information.

So, there are two things that need to be done.  The first is to get the DID into the accountcode and to do that, we simply add a new context to /etc/asterisk/extensions_custom.conf as follows:

[custom-from-trunk-accountcode]
exten => _X!,1,Set(CDR(accountcode)=${EXTEN})
exten => _X!,n,Goto(from-trunk,${EXTEN},1)
exten => h,1,Macro(hangupcall,)

NOTE: The above assumes that all your DID’s are entirely numeric.  It will fail (and calls may not reach your inbound routes) if you do not first strip off any non-numeric characters, or change the context to accept them.  For example, if a provider sends a + at the start of the DID (NOT the Caller ID, which is more common), then try changing both occurrences of _X! to _[+X]!

Then for each of our trunks, we go into FreePBX and change the context= line from context=from-trunk to:

context=custom-from-trunk-accountcode

If by some chance you are already sending calls from one or more of your trunks to a custom context in extensions_custom.conf rather than from-trunk, you can simply add one of the following lines to that custom context. If you add it as the first line, use:

exten => _X!,1,Set(CDR(accountcode)=${EXTEN})

And make sure that any subsequent line is not also numbered “1”.  If it’s NOT the first line, use:

exten => _X!,n,Set(CDR(accountcode)=${EXTEN})

That should get your DID into the accountcode field of /var/log/asterisk/cdr-csv/Master.csv

To actually display the field, I refer you to this article on the PSU VoIP blog:

Modify FreePBX call reports to show destination channel

Read that article and the comments underneath, because it explains the principle of displaying additional fields in the FreePBX CDR.  This is how I set up my /etc/asterisk/call-log-table.php file:

calldate", "10%", "center", "SORT", "19");
        $FG_TABLE_COL[]=array ("Source Channel", "channel", "14%", "center", "", "32");
        $FG_TABLE_COL[]=array ("Source DID", "accountcode", "10%", "center", "SORT", "20");
        $FG_TABLE_COL[]=array ("Source", "src", "10%", "center", "SORT", "30");
        $FG_TABLE_COL[]=array ("CLID", "clid", "20%", "center", "", "80",'','','','',
          '','filter_html');
        $FG_TABLE_COL[]=array ("Dest.", "dst", "10%", "center", "SORT", "30");
        $FG_TABLE_COL[]=array ("Dest. Channel", "dstchannel", "14%", "center", "", "32");
        $FG_TABLE_COL[]=array ("Disposition", "disposition", "6%", "center", "", "30");
        if ((!isset($resulttype)) || ($resulttype=="min"))
          $minute_function= "display_minute";
        $FG_TABLE_COL[]=array ("Duration", "duration", "6%", "center", "SORT", "30",
          "", "", "", "", "", "$minute_function");

        $FG_TABLE_DEFAULT_ORDER = "calldate";
        $FG_TABLE_DEFAULT_SENS = "DESC";

        // This Variable stores the argument for the SQL query
        $FG_COL_QUERY='calldate, channel, accountcode, src, clid, dst, dstchannel, disposition,
          duration';
        $FG_COL_QUERY_GRAPH='calldate, duration';

        // The variable LIMITE_DISPLAY define the limit of record to display by page
        $FG_LIMITE_DISPLAY=25;

        // Number of column in the html table
        $FG_NB_TABLE_COL=count($FG_TABLE_COL);

        // The variable $FG_EDITION define if you want process to the edition of the
        // database record
        $FG_EDITION=true;

        //This variable will store the total number of columns
        $FG_TOTAL_TABLE_COL = $FG_NB_TABLE_COL;
        if ((isset($FG_DELETION) && $FG_DELETION) || $FG_EDITION)
          $FG_TOTAL_TABLE_COL++;

        //This variable define the Title of the HTML table
        $FG_HTML_TABLE_TITLE=" - Call Logs - ";

        //This variable define the width of the HTML table
        $FG_HTML_TABLE_WIDTH="100%";
?>

As always you should copy and paste the code, so you get complete lines including anything that WordPress truncates on the display.

The above shows the Source DID column just to the right of the Source Channel column.  I also made Source DID a sortable column, so you can show your usage by DID by clicking on the “Source DID” column heading.

At the start I said that you can only use this if you’re not using Account Codes on your system.  Actually, that may not be strictly true.  A DID would normally only be saved for an incoming call, while an Account Code would normally only be saved for an outgoing call.  So, it may be possible to make that field do double duty, but since I don’t use Account Codes and don’t know anyone who does, I’ll leave that as an exercise for the reader.

Using a dynamic DNS (DDNS) to solve the problem of keeping a firewall open to remote users at changeable IP addresses

 

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.

(Updated July 1, 2011 to include rudimentary test for string returned that doesn’t contain an actual IP address)

One problem faced by Asterisk users (and probably also users of other software PBX’s) is that you want to secure your system by not opening ports up to the entire Internet, but if you have remote users (users not on the same local network as your Asterisk server) you need to make an exception for them to allow them to penetrate your firewall.  If all your external users have fixed IP addresses, it’s not a problem — you simply add a specific rule in your firewall to permit access from each user’s IP address.  However, if their ISP changes their IP address frequently, or if they are using a softphone on a laptop computer, then you can’t just assume they will constantly be at same IP.  And if one of those users happens to be your boss or your mother, they are not going to be happy if they can’t use the phone until they make contact with you, and you enter their new IP address in the firewall.  And they’re probably not going to be real happy if they have to go to a web site or take some other action before they can make and receive calls.

This solution will work for many users in this situation, provided that you are using the iptables firewall. Again, the goal is to keep all your ports closed to outsiders, except for your authorized users. But if you can get each user to set up a Dynamic DNS account and then set their router to do the Dynamic DNS updates (as described here for DD-WRT users), OR failing that if you can get them to install a software Dynamic DNS client on their computer (which is a poorer choice because the computer has to be on for updates to occur), then you can run a script on your Asterisk box every five minutes to check to see if their IP address has changed, and if so, update iptables. I have one script that is called as a cron job every five minutes, and looks like this:

#!/bin/bash
/root/firewall-dynhosts.sh someaddress.afraid.org
/root/firewall-dynhosts.sh someotheraddress.afraid.org
/root/firewall-dynhosts.sh someaddress.no-ip.com

In other words it has one line for each Dynamic DNS host I want to check. For each host it calls a script named firewall-dynhosts.sh which in turn contains this:

#!/bin/bash
# filename: firewall-dynhosts.sh
#
# A script to update iptable records for dynamic dns hosts.
# Written by: Dave Horner (http://dave.thehorners.com)
# Released into public domain.
#
# Run this script in your cron table to update ips.
#
# You might want to put all your dynamic hosts in a sep. chain.
# That way you can easily see what dynamic hosts are trusted.
#
# create the chain in iptables.
# /sbin/iptables -N dynamichosts
# insert the chain into the input chain @ the head of the list.
# /sbin/iptables -I INPUT 1 -j dynamichosts
# flush all the rules in the chain
# /sbin/iptables -F dynamichosts

HOST=$1
HOSTFILE=”/root/dynhosts/host-$HOST”
CHAIN=”dynamichosts” # change this to whatever chain you want.
IPTABLES=”/sbin/iptables”

# check to make sure we have enough args passed.
if [ “${#@}” -ne “1” ]; then
echo “$0 hostname”
echo “You must supply a hostname to update in iptables.”
exit
fi

# lookup host name from dns tables
IP=`/usr/bin/dig +short $HOST | /usr/bin/tail -n 1`
if [ “${#IP}” = “0” ]; then
echo “Couldn’t lookup hostname for $HOST, failed.”
exit
fi

if [ ! `expr “$IP” : ‘([1-9])’` ]; then
echo “Did not return valid IP address, failed.”
exit
fi

OLDIP=””
if [ -a $HOSTFILE ]; then
OLDIP=`cat $HOSTFILE`
# echo “CAT returned: $?”
fi

# has address changed?
if [ “$OLDIP” == “$IP” ]; then
echo “Old and new IP addresses match.”
exit
fi

# save off new ip.
echo $IP>$HOSTFILE

echo “Updating $HOST in iptables.”
if [ “${#OLDIP}” != “0” ]; then
echo “Removing old rule ($OLDIP)”
`$IPTABLES -D $CHAIN -s $OLDIP/32 -j ACCEPT`
fi
echo “Inserting new rule ($IP)”
`$IPTABLES -A $CHAIN -s $IP/32 -j ACCEPT`

echo “Changing rule in /etc/sysconfig/iptables”
sed -i “0,/-A\sdynamichosts\s-s\s$OLDIP\s-j\sACCEPT/s//-A dynamichosts -s $IP -j ACCEPT/” /etc/sysconfig/iptables
# sed -i “s/-A\sdynamichosts\s-s\s$OLDIP\s-j\sACCEPT/-A dynamichosts -s $IP -j ACCEPT/g” /etc/sysconfig/iptables

echo “Sending e-mail notification”
`echo “This is an automated message – please do not reply. The address of dynamic host $HOST has been changed from $OLDIP to $IP. You may need to change the dynamichosts chain in Webmin’s Linux Firewall configuration.” | mail -s “IP address of dynamic host changed on machine name recipient@someaddress.com,anotherrecipient@someaddress.net`

As always, copy and paste the above script, so you can see where the line breaks are really supposed to be (the last line in particular is quite long, and will likely be broken up into four or five lines on the screen). Also, beware of WordPress or other software changing the single or double quotation marks to “prettified” versions — only the plain text normal quotation marks will work.

Note that prior to the first run of the script you will need to run the three commented-out commands shown near the top of the script, right after “create the chain in iptables”, to create the chain. For your convenience here they all are in one place, without the interleaved comment lines:

/sbin/iptables -N dynamichosts
/sbin/iptables -I INPUT 1 -j dynamichosts
/sbin/iptables -F dynamichosts

The lines in blue in firewall-dynhosts.sh are custom additions by me. Just in case something goes wrong, I suggest you make a backup copy of /etc/sysconfig/iptables in a safe place before running this script.  My first addition checks the first character of the string returned in $IP to make sure it is actually a number.  This was a quick and dirty addition to keep it from trying to use a string like ;; connection timed out; no servers could be reached as a valid IP address (yes, it really did that).  I’m sure that the test there could be improved upon (for example, to do a full check for a valid IP address rather than just checking the first digit) but as I say this was a quick and dirty fix.  If you have any suggestions on how to improve it, please leave a comment.  I did find this article, Validating an IP Address in a Bash Script, but it seemed like a bit of overkill considering that in this case what I’m really trying to do is simply weed out error messages.

The second set of additions change the address in the dynamichosts chain of /etc/sysconfig/iptables. Please note that this file may be at a different location in some versions of Linux (such as /etc/iptables.up.rules), if so you will need to change this accordingly. This is particularly important if you run both Webmin and fail2ban. If fail2ban is running it will add some lines to the in-memory version of iptables, so you don’t want to do a simple commit to save the in-memory version back to the iptables file. But at the same time, if you use Webmin’s “Linux Firewall” module to maintain iptables, you want any changes in IP addresses to show up the next time you call up Webmin’s Linux Firewall page. So this simply does a search and replace in /etc/sysconfig/iptables on the rule containing the old IP address, and replaces it with the new one. There are two lines in that section that contain the sed command, the first one will replace only the first instance of the old IP address if it’s in iptables more than once, while the second (which is commented out) would replace all instances of the old IP address. Uncomment whichever you prefer and leave the other commented out, but bear in mind that if two or more of your remote extensions might ever be at the same IP address at the same time, you want the first version (the one that is uncommented above) so that when one of those extensions moves to a different IP address it doesn’t change the IP address for all of the extensions.

Note there’s still a possibility of missing a change if you are actually working in Webmin when a change occurs (since you’ll already have loaded a copy of iptables, and if you then make changes and save it out it could overwrite any change made by the script). But, the last two lines of the script send you an e-mail to alert you to that possibility. If you don’t use Webmin and don’t need or want an e-mail notification for some other reason, you can omit those last two lines, otherwise change the parts in red text to sane values for your situation. While editing, pay attention to the backtick at the end of the line (it’s easy to accidentally delete it when editing an e-mail address — don’t do that!).

When you’re all finished, make sure both scripts are executable and the permissions are correct, then create a cron job to call the first script every five minutes.

The only slight drawback to this method is that when an IP address changes it can take up to ten minutes to update (five for the Dynamic DNS to pick it up, and five more for the cron job to fire that gets it from the Dynamic DNS). Fortunately, most ISP’s tend to change IP address assignments in the middle of the night. Note that using the wrong DNS servers can cause the updates to take significantly longer; I set my computers to use Google’s DNS (8.8.8.8 and 8.8.4.4) and that works fairly well. Note that if ALL your Dynamic DNS addresses are from freedns.afraid.org then you may want to change one line in the above script, from

IP=`/usr/bin/dig +short $HOST | /usr/bin/tail -n 1`

to

IP=`/usr/bin/dig +short @ns1.afraid.org $HOST | /usr/bin/tail -n 1`

This change will specify that the afraid.org DNS server is to be used for these lookups (and ONLY for these lookups, not for every DNS request your system makes – don’t want to overload the servers of this free service!). This may be particularly important if the DNS server you normally use is a caching server that doesn’t always do real-time lookups for each DNS request (for example, if you have installed the BIND DNS Server on your system). If some of the Dynamic DNS addresses come from other services then you could use a similar modification that checks a public DNS service that does not cache entries for long periods of time; as I write this Google’s DNS servers seem to update in near real time.

One thing some may not like is that this script basically hands the “keys to the kingdom” to your authorized users, by giving them access to all ports, or at least all ports not explicitly denied by rules higher in priority. It would be easy enough to change the rule that is written to iptables, or even add additional ones, in the above script, so that you could specify access to individual ports. The other problem is it works great for those external users at fixed locations that don’t move around a lot. It might not work quite as well as well for softphone users on laptops due to the delay between the time they turn on the laptop and the time your Asterisk server picks up the new address.

This has actually worked the best for me of anything I’ve tried so far because once you get the external user’s router set up to do the Dynamic DNS updates, they don’t have to think about doing anything else prior to making a call.

EDIT (December, 2015): If it is not possible or appropriate to update the dynamic DNS automatically from the users’ router, there may be another option. If any of your users have Obihai devices (or possibly another brand of VoIP device that includes an accessible “Auto Provisioning” feature that is not currently being utilized), you may want to know that they do not need to run a separate client to update their dynu.com or freedns.afraid.org dynamic IP address, because an Obihai device (and possibly some other brands of VoIP devices) can do that automatically. This is NOT a recommendation for Obihai devices, but if you or one of your users happens to already have one, here is the information as originally found in this thread on the Obihai forum, posted by user giqcass, who wrote:

Rough Draft for hackish DNS updates:

This hack will let your OBi update Dynamic DNS. It isn’t perfect but it works very well. It’s as simple as calling a url to update the DNS at afraid.org. I believe it would be a simple task to add this feature to the OBi firmware directly. So please add this OBiHai. Pretty please. Until then here you go.

Set up a Dynamic DNS host at http://freedns.afraid.org/
Go to the Dynamic DNS tab.
Copy the “direct” update url link.
Open your Obi admin page.
Click the System management page.
Click Auto Provisioning.
Under “ITSP Provisioning” Change the following.
Method = Periodically
Interval = This setting must be greater then 400 so not to over use resources. I use 3667.
ConfigURL = Paste the update link you got from afraid.org (use http://… not https://…)

Press Submit at the bottom of the page. Restart you OBi.

If you use choose to use dynu.com instead of freedns.afraid.org (which you might because dynu.com doesn’t force you to visit their web site periodically to keep your domain), the procedure is the same (after the first line), except that for the ConfigURL you would use:

http://api.dynu.com/nic/update?hostname=YOUR_DYNU_DYNAMIC_DNS&username=YOUR_DYNU_USERNAME&password=MD5_HASH_OF_PASSWORD

Replace YOUR_DYNU_DYNAMIC_DNS with your dynamic DNS domain name, YOUR_DYNU_USERNAME with the username you use to log into your dynu.com account, and MD5_HASH_OF_PASSWORD with the MD5 hash of your dynu.com password OR your IP Update Password if you have set one (which is recommended). To get the MD5 hash of the password you can enter it on this page. To set or update your IP Update Password, use this page.

The advantage of this is that if one of your users travels and takes their VoIP device with them, it would be able to change the dynamic DNS each time they plug in at a new location (not immediately, but after several minutes at most), so that if you use the technique outlined in this article your server will recognize their current address and permit access. Remember that it’s okay to use more than one Dynamic DNS service simultaneously, in case you or your user are already using a different one that doesn’t provide a simple update URL like dynu.com and freedns.afraid.org do. Other brands of VoIP adapters that have a similar “Auto Provisioning” feature may be able to do this as well, but we don’t have specific information for any of them. If you do, please feel free to add that information in a comment.

Note that we are not recommending any particular free dynamic DNS service. If you want to know what your options are, there is an article on the Best Free Dynamic DNS Services that will show you some options. You want one that is reliable and that will not disappear in a few months, but since we don’t have a crystal ball, we can’t tell you which ones might fit that criteria.

Do you use Webmin to configure iptables and also run fail2ban? Don’t forget to do this!

 

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.

For many Linux users this will be a “Thank you, Captain Obvious” type of post, but it’s one of those things that some Webmin users might not realize.  If you use Webmin’s “Linux Firewall” configuration page to configure the iptables firewall in Linux, and you click “Apply Configuration”, it will remove fail2ban‘s rules from your active iptables configuration.  So, you must go to a Linux command prompt and enter service fail2ban restart — UNLESS you make a small change in the Webmin “Linux Firewall” configuration.

From the Webmin “Linux Firewall” main page, click Module Config, then on the configuration page, in the “Configurable options” section, look for the line “Command to run after applying configuration.” Click the button next to the text box on that line, and in the text box enter service fail2ban restart and then click the Save button at the bottom of the page. That’s all you need — now every time you make a firewall change and click “Apply Configuration”, it will automatically restart fail2ban for you.

A Perl script to send Caller ID popups from Asterisk to computers running Notify OSD (such as Ubuntu Linux) or any command-line invoked notification system

 

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 is basically an update to my article, A Perl script to send Caller ID popups from Asterisk to computers running Growl under OS X on a Mac or Growl for Windows, and you should still use that article if you are sending notifications to a computer on your local network that runs Growl or Growl for Windows as the notification system.

I wanted to find a way to send Caller ID popups to a Ubuntu Linux box, and in the process I discovered a different method of sending such notifications.  There are pros and cons to using the new method, so let me explain those first:

Pros:

  • Can send notifications to any computer that supports command line generated notifications (so it could also be used with Growl, if you can use growlnotify from a command prompt to generate a notification).
  • Can send notifications to any computer that you can SSH into, provided you have it set up to use public/private key authentication rather than password authentication.

Cons:

  • Notifications typically display a couple of seconds later than under the previous method.  I suspect this is due to the SSH authentication taking a second or two.
  • It’s a little bit more complicated to set this up, though not horribly so.
  • Because this uses SSH and requires that Asterisk be granted permission to establish an SSH connection as the super user (by using sudo), there may be unforeseen security risks.

Read that last point again, and please understand that as with all projects on this site, I offer this for experimental purposes only.  I explicitly do not warrant this method as being 100% secure, nor will I tell you that it could not be exploited to do bad things on your system.  I don’t think it can (and feel free to leave a comment if you think I’m wrong), but I just don’t know that for sure.  So, if you decide to use anything in this article, you agree to assume all risks. If you’re the type that likes to sue other people when something goes wrong, then you do not have permission to use this code.  We’re all experimenters here, so no guarantees!

As with the previous method, you must have the Perl language installed on your Asterisk server, and you must have the Asterisk::AGI module 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.

There’s one additional thing you must do on the Asterisk server before this will run, and that’s allow Asterisk to run the ssh command as root. So, add this to your /etc/sudoers file (probably at the very end, but in any case it should be obvious where to add this because it will be in a section where Asterisk is granted similar privileges with regard to other programs):

asterisk ALL = NOPASSWD: /usr/bin/ssh

Next you want to copy and paste the following Perl script to the filename /var/lib/asterisk/agi-bin/notifysend.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 have modified.

#!/usr/bin/perl
use strict;
use warnings;
use Asterisk::AGI;
my $agi = new Asterisk::AGI;
my %input = $agi->ReadParse();

# Next two lines fork the process so Asterisk can get on with handling the call
open STDOUT, '>/dev/null';
fork and exit;

my $num = $input{'callerid'};
my $name = $input{'calleridname'};
my $ext = $input{'extension'};
my $user = $ARGV[0];
my $ip = $ARGV[1];

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

# OMIT this section if you don't want IP address
# checking (e.g. you want to use foo.bar.com)
unless ( $ip =~ /^(d+).(d+).(d+).(d+)$/ ) {
    exit;
}

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

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

my $cmd = qq(./remotenotify.sh "$name" "$num calling $ext at $fulldate");
$cmd = "sudo ssh $user@$ip '$cmd'";
exec "$cmd";

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 (if you have not already done so when using the previous method). Note that if you have a mix of computers on your network and you are using both the new and old methods, you only need to do this once — it works with both methods (hence the reference to “growlsend” in the database and “gshelper” as the name of this script). Call it /var/lib/asterisk/agi-bin/gshelper.agi and note that there is a line within it that you may need to change to reflect the scope of your local network:

#!/usr/bin/perl
use strict;
use warnings;
my ($prev, @mac, @ip);
# Change the 192.168.0.0/24 in the following line to reflect the scope of your local network, if necessary
my @nmap = `nmap -sP 192.168.0.0/24|grep -B 1 MAC`;
foreach (@nmap) {
    if (index($_, "MAC Address:") >= 0) {
        @mac = split(" ");
        @ip = split(" ",$prev);
        `/usr/sbin/asterisk -rx "database put growlsend $mac[2] $ip[1]"`;
    }
    $prev=$_;
}

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 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). However, note that the syntax used in extensions_custom.conf differs from what is shown there, depending on whether you are specifying an IP address or a MAC address to identify the target computer.

First, if you are specifying the IP address of the target computer, then instead of using this syntax:

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

You will need to use this:

exten => ****525,1,AGI(notifysend.agi,username,192.168.0.123,525)

Note that username is the account name you use when doing an ssh login into the destination system, and it should also be the desktop user on the system (not root!). Let’s say that the system is currently at IP address 192.168.0.123. In order for this to work, you need to be able to ssh into your Ubuntu box from your Asterisk server, using the following command from the Asterisk server’s command line:

ssh username@192.168.0.123

If it asks for a password, then you need to follow the instructions at Stop entering passwords: How to set up ssh public/private key authentication for connections to a remote server, and get it set up so that it will not ask for a password (if you don’t like my article, maybe this one will make it clearer).

It’s probably easiest to configure each computer that is to receive notifications to use a static IP address. 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(notifysend.agi,username,01:23:45:AB:CD:EF,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 (from gshelper.agi) 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(notifysend.agi,username,192.168.0.123,525)

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.

Now is where it gets just a bit more complicated than in the original method. If you have followed the above instructions, you’ll be able to send the notifications to the remote system using SSH, but there will be nothing there to receive them. So we have to create a small script on the receiving system to do something with the received notifications. That script will vary depending on the receiving system, but it must be named remotenotify.sh and it must be placed in the destination user’s home directory, and don’t forget to make it executable! Here’s one that will work in most Ubuntu installations that have Notify OSD installed:

export DISPLAY=:0
notify-send --urgency="critical" --icon="phone" "$1" "$2"

Those two lines are all you need. On a different type of system (or if you have multiple displays) you may need to or wish to do something different. For example, as I mentioned above, if the destination system is running Growl then your remotenotify.sh script will need to call growlnotify, but beyond that I wouldn’t know what to use there (EDIT: But if the target system is a Mac that is running OS X, a pretty good guess would probably be that you’d only need one line, something like this:

growlnotify -s -p 1 -a Telephone -m "$2" $1

In this case it should make the notification sticky until dismissed by the user, give it a priority of 1 — the default is 0 — and use the application icon from the “Telephone” application if you have it installed. Instead of -a to specify an application’s icon you could use -I followed by a path to an .icns file that contains an icon you want to use.  Type growlnotify –help to see all the growlnotify options.  Oh, and before you can make an SSH connection to a Mac you have to go into System Preferences | Sharing and turn on Remote Login).

The beauty of this approach is that you can make the remotenotify.sh script as simple or as complicated as you need — you could even make it forward a notification to other devices if you wish, but figuring out how to do that is up to you (if you come up with something good, please leave a comment and tell us about it!).

If you’re running Ubuntu on the target system, here’s a few articles you may wish to use to help you get your notifications to look the way you want them to appear:

Tweak The NotifyOSD Notifications In Ubuntu 10.10 Maverick Meerkat [Patched NotifyOSD PPA Updated]
Get Notifications With A Close Button In Ubuntu
Configurable NotifyOSD Bubbles For Ubuntu 11.04: Move, Close On Click, Change Colors And More

If you want to be able to review missed notifications, you may be able to use this (as a side note, why don’t they have something like this for Growl?):

Never Miss A NotifyOSD Notification With “Recent Notifications” GNOME Applet

The idea behind the shell script that runs on the target system was found in a comment on the following article, which may be of special interest to MythTV users:

Send OSD notification messages to all systems on a network

There are links to other original sources throughout the article, so feel free to follow those if you want more in-depth commentary.

Link: Monit: Disk space monitoring

 

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 February, 2011 and may be out of date. You may also wish to read Link: How to Install and Setup Monit (Linux Process and Services Monitoring) Program.

A hard disk drive with the platters and motor ...
Image via Wikipedia

Here’s an article that will be helpful to may of you who are running PBX servers under CentOS, especially (but not limited to) those running on virtual machines with low disk storage space.  Note that if you installed from an “all-in-one” distribution ISO, this possibly might already be installed, but may still need to be configured.

One thing you definitely don’t want to happen to your server is for it to run out of disk space, especially the root partition.

There are lots of pieces of open source monitoring software, a popular one being monit.

Below is a quick guide to installing monit and generating alert e-mails for disk space and cpu/memory usage. The installation was done on a SysAdminMan VPS running CentOS 5.5

Full article here (SysAdminMan)

The instructions should work for any system running CentOS 5.5.  You might be tempted to take a shortcut and just do “yum install monit” but please be aware that (at least as of the day I’m writing this) it will get you a much older version of the software, so I suggest you stick with the instructions in the article.  I have just now installed this on one system and have not fully tested it, but it did send an e-mail confirming that it had started.

This is just another tool you can use to make your life a little easier and help you avoid a problem before it becomes a major headache!

FreePBX voicemail hacks

 

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 January, 2011 and may contain outdated information.

Motel Phone
Image by Andreas_MB via Flickr

A few things you should know about FreePBX voicemail:

• If you are not receiving voicemail notifications (stutter dial tone, message waiting indication on certain phones, etc.) there are two things to check.  One is to go into /etc/asterisk/vm_general.inc and see if there is a line of the form ;pollmailboxes=yes — if it is commented out (semicolon in front), uncomment it by removing the semicolon.  The other is to go to the /var/spool/asterisk/voicemail directory and make sure that you have directories there called default and device, and that one is symlinked to the other (generally default is the “real” directory and device is the symlink). If the device directory is missing, make sure you’re in the voicemail directory and do this:  ln -s default device

• If you have users in different time zones, you can have the voicemail “envelope” information say the correct time by creating a [zonemessages] context at the end of /etc/asterisk/vm_general.inc (in later versions of FreePBX you can also enter these in the Timezone Definitions section of the Voicemail Admin module) — here’s a simple one showing the four major time zones in the continental United States (I know this is not complete, it’s just an example):

[zonemessages] eastern=America/New_York|'vm-received' Q 'digits/at' IMp
central=America/Chicago|'vm-received' Q 'digits/at' IMp
mountain=America/Denver|'vm-received' Q 'digits/at' IMp
pacific=America/Los_Angeles|'vm-received' Q 'digits/at' IMp

Then, on each extension setup page in FreePBX, find the Voicemail & Directory section, and under that the VM Options.  In the VM Options add a tz= option for each user (for example, tz=eastern), using one of the zones you defined under [zonemessages] in /etc/asterisk/vm_general.inc.  Note that multiple options in VM Options must be separated by the | (vertical bar) character (not that you’re likely to have multiple options, but I mention it just in case).

• If your system is not used in a large office, or some other location where not all users can be trusted, you can disable the requirement to enter a PIN when using *97 to pickup voicemail for your own extension.  To do that, add the following context to /etc/asterisk/extensions_custom.conf:

NOTE: This is the original code for older versions of FreePBX:

[custom-voicemail-retrieve] exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Macro(user-callerid,)
exten => s,n,Macro(get-vmcontext,${CALLERID(num)})
exten => s,n,VoiceMailMain(${CALLERID(num)}@${VMCONTEXT},s)
exten => s,n,Macro(hangupcall,)
exten => h,1,Macro(hangupcall,)

In newer versions of FreePBX (probably 2.9 and later) use this instead:

[custom-voicemail-retrieve] exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Macro(user-callerid,)
exten => s,n,Macro(get-vmcontext,${AMPUSER})
exten => s,n(check),Set(VMBOXEXISTSSTATUS=${IF(${MAILBOX_EXISTS(${AMPUSER}@${VMCONTEXT})}?SUCCESS:FAILED)})
exten => s,n,GotoIf($["${VMBOXEXISTSSTATUS}" = "SUCCESS"]?mbexist)
exten => s,n,VoiceMailMain()
exten => s,n,GotoIf($["${IVR_RETVM}" = "RETURN" & "${IVR_CONTEXT}" != ""]?playret)
exten => s,n,Macro(hangupcall,)
exten => s,check+101(mbexist),VoiceMailMain(${AMPUSER}@${VMCONTEXT},s)
exten => s,n,GotoIf($["${IVR_RETVM}" = "RETURN" & "${IVR_CONTEXT}" != ""]?playret)
exten => s,n,Macro(hangupcall,)
exten => s,n(playret),Playback(beep&you-will-be-transfered-menu&silence/1)
exten => s,n,Goto(${IVR_CONTEXT},return,1)
exten => h,1,Macro(hangupcall,)

Then do the following in FreePBX’s GUI (do these steps in the order shown):

Go to Feature Codes and under Voicemail, disable “My Voicemail” (*97) using the dropdown, then Submit Changes.

Go to Custom Destinations (under the Tools tab) and create a new Custom Destination:  custom-voicemail-retrieve,s,1 — then Submit Changes.

Go to Misc. Applications and add a new Misc. Application. Make the feature code *97 and the destination the Custom Destination you created in the previous step, then Submit Changes.

Finally do an “orange bar reload” in FreePBX. Now when your users dial *97, it will assume they are authorized to pick up the voicemail for the extension they’re calling from. Obviously, this is probably not a good idea in any kind of office setting.

Got any other FreePBX voicemail hacks you like?

Link: Interesting security technique for Asterisk and FreePBX users (may work with other SIP-based PBX’s also)

 

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 November, 2010.

NOTE: For some reason WordPress absolutely hates it when I try to edit this post, and turns links and other things into piles of steaming poo.  If things don’t look right here please e-mail me or leave a comment and I’ll check it out.  WordPress, I KNOW how I want my articles to look, why can’t you just leave them alone?

One problem faced by some SIP-based VoIP PBX administrators is the issue of security when you have external extensions (that is, extensions located anywhere in the world that’s not a part of your local network). You want to allow those extensions (the ones you’ve authorized) to connect to your system, but you prefer to keep everyone else out, and preferably not even tip them off that there’s a PBX there. The idea is, if the bad guys that would like to break into PBX’s don’t even realize that there is a PBX at your IP address, they won’t waste any time trying to crack into your system.

There have been other suggestions for how to handle this but many of them require your users to take some additional action(s) that they would not normally have to take, and users hate having to lift a finger to do anything to enhance their security. Which brings us to a rather clever technique that doesn’t require user to do anything other than use their phones as they normally would. It might be a tiny bit of a pain to set up initially, but the results may be worth it. I would call this medium level security because if someone is sniffing your packets, this alone may not keep them out, but most of the lowlifes that try to break into PBX’s don’t actually have sufficient access to sniff your packet stream (and also, they’d have to know the exact technique you’re using to be able to crack this). So without further ado…

Secure your VoIP server with the SunshineNetworks knock

(As of October 24, 2012, the above link appears to be DEAD — see the edit at the end of this article)

Note that while the article recommends changing the SIP port to something other than 5060, their basic technique (the “knock”) should still work even if you feel you need to stay on 5060. My only fear about changing the SIP port would be the possibility of losing communications with VoIP providers and with other systems I legitimately send/receive voice traffic to/from. They’re probably going to keep using 5060 even if I don’t. EDIT: My concern here may be unfounded — note the comment below from Alex of Sunshine Networks, who said that “changing the SIP port is quite safe. Your SIP server will send this SIP port along in it’s first SIP invite registration to the VoIP provider. So unless your VoIP provider is actively blocking out anything else than port 5060, it should work fine. We use this technique with 3 different major SIP providers in Australia and never had problems. So far we haven’t seen any unintended consequences.”

I haven’t personally tested this, so if you do, please consider leaving a comment to let me know how it worked for you. The two things I wonder are, do these rules survive a reboot, and can you have more than one secret phrase that would let people in (in case you want to use a different one for each external extension)? EDIT: Those questions are also addressed in Alex’s comment below. Also, those of you running PBX in a Flash should take note of Ward Mundy’s comment about changing an entry in /etc/sysconfig/iptables in this thread. In that same thread, there appears a method to view the “knock” each extension is currently sending — just do “sip debug” from the Asterisk CLI for an hour or so (long enough for all your endpoints to register, after which you can use “sip no debug” to turn it off), then run this at the Linux command prompt (not from the CLI!):

grep "From: " /var/log/asterisk/full|cut -f1 --delimiter=; | sort -u

For each of your remote extensions, you’ll see a line that looks something like this:

From: The Knock <sip:234@nn.nn.nn.nn>

“The Knock” may or may not be enclosed in quotation marks, but it apparently doesn’t matter (you don’t include them in the iptables rules). If you haven’t used a specific “knock”, it could be the actual user’s name, if you set that up when you first set up the endpoint. Anyway, I’d suggest running this BEFORE you actually implement the iptables rules, so you know ahead of time what each endpoint is sending.

EDIT (Added January 8, 2012): I am now using a slight variation on this technique on one of the systems I administer. Without going into too many specifics, I will just note that some SIP devices and VoIP adapters actually already send a unique string that you can use as a “knock” – you do not have to configure a new one, you just need to find out what the device is already sending and use that. For example, let’s say you have an VoIP device connecting to your Asterisk server as extension 234. All you have to do is go to the Asterisk CLI (NOT the Linux command prompt) and enter this:

sip set debug peer 234

(Replace 234 with the actual extension number). Now, assuming that the device is connecting to your server, you will start to see SIP packets scroll across your screen. Within a few minutes you should see one like this (IP addresses have been xx’ed out):

<--- SIP read from UDP:xx.xx.xx.xx:5061 --->

REGISTER sip:xx.xx.xx.xx:5060 SIP/2.0
Call-ID: e10700c2@xx.xx.xx.xx
Content-Length: 0
CSeq: 56790 REGISTER
From: <sip:234@xx.xx.xx.xx>;tag=SP8f427e45f1e19cb24
Max-Forwards: 70
To: <sip:234@xx.xx.xx.xx>
Via: SIP/2.0/UDP xx.xx.xx.xx:5061;branch=z4b9hGK-4f0473a8;rport
Authorization: DIGEST algorithm=MD5,nonce=”37cd169d”,realm=”asterisk”,response=”a726bfed5db321a7bc967b997b5157c2″,uri=”sip:xx.xx.xx.xx:5060″,username=”234″
User-Agent: xxxxxx/xxxxxx-x.x.x.x
Contact: <sip:234@xx.xx.xx.xx:5061>;expires=60;+sip.instance=”<urn:uuid:nnnnnnnn-nnnn-nnnn-nnnn-nnnnnnnnnnnn>”
Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER
Supported: replaces

<————->

If you don’t see this you may need to increase the debug level. After you see a packet like this, you can turn off sip debugging:

sip set debug off

The string you are looking for is in the Contact: string above (the nnnnnnnn-nnnn-nnnn-nnnn-nnnnnnnnnnnn is replaced by a unique string). So, where in the instructions for the “knock” they show a sample string such as:

iptables -I door 1 -p udp --dport 5060 -m string --string "mysecretpass" --algo bm -m recent --set --name portisnowopen

I would change the --dport parameter to 5060:5061 (since an VoIP adapter sometimes uses port 5061 for the second service provider — for an device that allows up to fours service providers, use 5060:5063) and the --string parameter to “<urn:uuid:nnnnnnnn-nnnn-nnnn-nnnn-nnnnnnnnnnnn>”, but using the actual string sent by the device, of course. I know the Sunshine Network people recommend using something other than port 5060 but I just can’t bring myself to go quite that far, and even their examples show 5060.

Some other SIP-compliant devices also send unique strings in their REGISTER packets. One that does NOT do so, as far as I am aware, is the venerable Linksys PAP2. And I also do not believe that any of the Sipura line of devices send such a unique string.

Naturally, if an intruder KNOWS you are using that technique, they could try a brute-force attack on the unique string. So I recommend only using this with “uncommon” extension numbers (not 200 or 1000, for example) and with a VERY strong secret/password on the SIP connection. But it is another line of defense against would-be intruders!

EDIT (Added October 24, 2012): The original article, and most of the original site for that matter, seems to have gone offline. While I’m not going to repost the original article here without permission, I will give you a few more details and a couple of excerpts. First, they advised that you change the SIP port to something other than 5060 – they suggested using something in the range 20001 through 49000, though I am not sure why. They uses port 34122 in their examples, and noted that if you are running PBX software that has a “SIP Settings” module, if your find a setting for “Bind Port”, that would be the one to change. Of course if you do this, you then have to change the SIP port on ALL your SIP-based phones and VoIP adapters.

With regard to the “knock” itself, they said this:

Technical information :
… Technically, our knock consists of a secret passphrase which is sent together with the first SIP packet from the phone to the server. SIP packets are text files, very much readable like http packets are. The SIP headers in a REGISTER invite packet have a lot of information, and one of those headers is called the “Display Name”. This display name is used only internally in your Asterisk server and has no other use, so we figured we could fill in anything and the Asterisk functionality would still work fine. We decided to use it as a port knock password.

How does it work :
The Asterisk administrator sets up a simple iptables rule. The iptables rule checks for a secret phrase inside packets sent to the SIP port ( 5060 by default, 34122 after having changed it ). Unless it finds this secret phrase, it will drop the packets to this port. All the remote phone has to do is fill in the “User Name” SIP property on his SIP phone with the secret phrase, and he will be able to connect.

What you then needed to do was to go to into your Asterisk server and from a Linux command prompt, issue the following command:

iptables -N door

Then for EACH “knock” string you want to use, you would do this from the command prompt (note this is only one line, and note that 34122 is the example port and “mykn0ckstr1ng” is an example “knock”):

iptables -I door 1 -p udp –dport 34122 -m string –string “mykn0ckstr1ng” –algo bm -m recent –set –name portisnowopen

If you have anyone that needs to register with your server but cannot send the “knock”, but is at a fixed IP address, you’d add a line like this for each instance (again the port and ip address would probably need to be changed, and note that an entire subnet can be specified as in this example — just leave off the /24 if it’s a single ip address):

iptables -A INPUT -p udp –dport 34122 –source 10.10.1.0/24 -j ACCEPT

Then you would enter these three lines, but again using the correct port rather than 34122. In the first line you see the number 4000 — that is amount of time in seconds that the port will be open, and should be greater than 3600 because that’s the default registration timeout for many sip phones and VoIP adapters. The original article notes that you could use 86400, which is a full day:

iptables -A INPUT -p udp –dport 34122 -m recent –rcheck –seconds 4000 –name portisnowopen -j ACCEPT
iptables -A INPUT -p udp –dport 34122 -j door
iptables -A INPUT -p udp –dport 34122 -j DROP

And finally, to make iptables use these rules, you’d enter:

service iptables save

The original Sunshine Networks article notes that…

This code keeps port 34122 closed ( DROP ) unless someone has opened the door ( door ) in which case they are allowed to pass the door for a little more than 1 hour ( 4000 seconds ). Each time the phone re-registers , the SIP secret pass header is sent, and the door is reopened for 4000 seconds. Since the default SIP reregistration time on many phones is 3600, the 4000 seconds will make sure that as long as the phone is connected to the SIP server, or needs to be connected, the dynamic firewall rule is always active.

Once you have done this, if you configure the Display Name or User Name setting with the “knock” string, it should be able to get through your firewall. Any phone that doesn’t have this string won’t. Of course you can always make the “knock” something that a phone already sends (in a SIP register packet), as noted in the previous edit, and then you don’t have to reconfigure the phone at all. If a phone or device tries to connect without sending the “knock”, the firewall won’t allow it (assuming you haven’t previously created some other rule that allows the traffic to pass) and the connection will fail, or at least that is how it’s supposed to work (I make no guarantees because I didn’t come up with this).

If you enter the command cat /proc/net/ipt_recent/portisnowopen you will get a list of IP addresses that have successfully used the “knock” to connect. Remember that after you implement this, it can take up to an hour for a device to attempt to reconnect.

If anyone ever spots the original article back online, please let me know and I’ll remove this edit. I’d rather you get the information direct from the original source anyway, and the short excerpts I have provided here don’t give the complete overview that the original article provided.

EDIT (February 23, 2014): It appears that there is an archived copy of that original article on the Wayback Machine, although we do not know if it is the most recent edit of that article prior to the site disappearing.

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