Category: FusionPBX

Asterisk on a Raspberry Pi – which distribution is best?

Portions of this article were UPDATED July 20, 2016, mostly to include information about Raspivo.

To the best of our knowledge there are five projects that will allow you to run a PBX on a Raspberry Pi. They are:

In this discussion we are only going to consider the first four, because FusionPBX runs on top of FreeSWITCH, not Asterisk. And we have nothing against FreeSWITCH, but it’s never been big among home users and experimenters. Perhaps that should change, but for now we just want to consider the Asterisk-based distributions.

It does not seem as though µElastix ever really caught on with a significant group of English-speaking users, and therefore it would be difficult to offer any sort of opinion. But we will note that new users and those not all that familiar with Linux may have a bit more trouble with the installation process, since there is no image file provided as is the case for some other distributions. One potential advantage of µElastix is that it will run on a Raspberry Pi, PicoSam, or Mcuzone, though you are not likely to run into the latter two boards anywhere in North America.

As for Incredible PBX, this takes the typical Nerd Vittles/PBX in a Flash “throw in everything but the kitchen sink” approach, but then offers this ominous-sounding advice:

Here’s everything you need to know about security for Incredible Pi:


What this basically means is that you can’t have any off-site extensions that register with your Asterisk server, if you heed their warning.  Well, you CAN, but not in any way that’s convenient for end users.  The problem apparently is that a few years ago someone connected with that project got hold of an article or two where someone got a huge phone bill by having an unsecured PBX, and had a major freakout about it.  There were probably several security failures associated with those incidents, but here is our question:  Since nobody in ANY other PBX project we’ve ever encountered gives advice like this, does this mean that Incredible PBX is incredibly insecure by design, and the only way to properly secure it is to take extraordinary steps such as these?

We’re not saying that all of this advice is out of line – the first point is probably a very good idea whenever possible – but most home users will be doing that anyway.  But it’s we particularly take issue with.  If you want to have any external extensions, you pretty much need to forward UDP ports 5060 and 10000-20000 to your Asterisk server.  And the Incredible PBX people specifically tell you not to do that, rather than recognizing that for some users that is simply not a viable option.

The other issue we have with Incredible PBX is that it includes a lot of what we would consider frivolous add-ons.  The main reason people generally install a PBX is because they want to use it to make phone calls, and perform a few other basic functions such as record voicemail, let callers select a destination from an auto-attendant, and so on.  All of these basic functions are provided by FreePBX, and all the other add-ons are pretty much useless unless you are just installing a PBX to play with features.  We can just about guarantee you that 99 percent of your users will not care that they can dial a code and get tide reports, or some similar nonsense.  On a regular server that has a lot of CPU power and storage space, having a bunch of extras may not be a problem.  On a Raspberry Pi, however, you are probably going to want a lean, trim installation that doesn’t get in the way of the basic functionality of a PBX.

I’ve seen reports in mid-2016 that Incredible PBX will soon (and perhaps already does) offer a menu at installation where you can select which features you want. However you will need to choose carefully because if you reject an option and then later decide you want it, you might need to reinstall from scratch to get it. New users might not know which features are actually useful and which are needless bloat, but at least it appears some effort is underway to stop forcing users to take all or nothing.

Raspivo is based on XiVO, which has been around for a while but was relatively unknown in the English-speaking world until fairly recently. However it has generated a lot of interest due to users desiring an alternative to FreePBX, which seems to be getting less “free” (in all senses of that word) as time goes by. There is a discussion about XiVO on DSLReports that you may wish to read, which in turn contains several other useful links. My understanding is that the “official” English language translation of the installation instructions are somewhat out of date, so you may find that RonR’s instructions on DSLReports are easier to follow. Like FreePBX, XiVO is a GUI interface for Asterisk, so any custom dialplan you have written for another build of Asterisk should be usable (perhaps with minor modification) in Raspivo. It appears that you must have at least a Raspberry Pi 2 or newer to run Raspivo.

If you have no experience at all with software PBX’s and are just getting your feet wet, but you are not unfamiliar with programming, XiVO is the one I’d suggest. It makes repetitive tasks easier but doesn’t get in your way when you want to customize your system to the same degree that FreePBX does. However if you just want everything to be as easy as possible, and you never want to do any dialplan customizations (or only very limited ones), then you may want to consider Asterisk for Raspberry Pi, also known as RasPBX.

The RasPBX distribution includes Asterisk and FreePBX, with additional scripts that will optionally let you install HylaFAX and/or Fail2Ban. There is also a related version of this software for the BeagleBone Black. This software is relatively easy to install, comes with no ominous security warnings, and doesn’t include a lot of “bloatware”, which we think is a definite advantage. They also have a semi-active discussion forum where you can find several installation and usage tips. And it is possible to run RasPBX from an External USB HDD or Thumb Drive, in case you are worried that running a PBX off of an SD card might not be reliable, although there are ways to minimize writes to the SD card if you prefer not to have the added power drain of another device.

We realize that none of these distributions are absolutely perfect, and everyone will have their own reasons for picking one over another. The PBX in a Flash forum used to include a Raspberry Pi board, but it was apparently lost in their “Great server crash of 2013”, and they never bothered to reinstate it. So it seems that for them, the Raspberry Pi is just one of many platforms they are attempting to support, and it does not appear to us that they are making much of an attempt to optimize their software specifically for the Raspberry Pi. We might receive a few less than gracious comments for saying that, but that’s simply our observation, and others are free to disagree – we just recall the old saying, “Jack of all trades, master of none” and feel it might apply in the case of putting out a version of Incredible PBX for the Raspberry Pi that includes pretty much everything that the versions intended for larger servers include. For performance reasons, we’d prefer to stick with a distribution designed for the Raspberry Pi from the ground up, and therefore our preference has always been Asterisk for Raspberry Pi / RasPBX, though nowadays we’d suggest that anyone that wants to have complete control over their system might also consider Raspivo. Just be aware that the learning curve with Raspivo might be a bit steeper.

If you disagree, feel free to try any of the other distributions mentioned. That’s the nice thing about having choices – you can try various programs until you find one that meets your needs, and maybe even your wants.

One final point – since this article was originally written in 2013, new versions of the Raspberry Pi have appeared, and some of the above-mentioned software may have been updated to only run on newer models. Or they may run, but only painfully slowly, if you have an original Raspberry Pi. In particular, it appears that Raspivo will only run on the Raspberry Pi 2 or newer. So if you have a first-generation Raspberry Pi, pay attention to the system requirements for the software you are downloading, because you might need to seek out an alternate or older version of the software.

Yes, you can run FusionPBX and FreeSWITCH on a Raspberry Pi


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.

By now most technically inclined folks have heard of the Raspberry Pi, the small $35 computer that can do big things. If you are going to buy one, just make sure you get one of the newer models with 512 MB of memory, rather than an older model with only 256 MB.

But, you may wonder, can I run a decent PBX system (one that won’t get in my way and treat me like a blithering idiot while I’m attempting to configure it) on a computer this small? Well, it turns out that people are doing just that:

The following guide is a relatively easy way to install FusionPBX and FreeSWITCH with the Ubuntu/Debian script.

Raspberry Pi Script (FusionPBX Wiki)

EDIT April, 2017: For a newer method see this DSLReports thread.

It should be obvious that you’ll probably find this easier if you know a bit about the Raspberry Pi first (Google it) but if you want a reliable and configurable PBX, and you think you have the skills to follow these instructions and make it work, I’d definitely give it a try. Besides, for home users, it’s a lot easier to justify a separate computer just to handle your phone calls if it’s small, cheap, and unobtrusive, and has low power consumption.

Using YATE to overcome Google Voice issues in FreeSWITCH and Asterisk


(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. The information in this article is VERY outdated and probably will not work.


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 have been less than thrilled with the Google Voice support in another software PBX, such as Asterisk or FreeSWITCH, you could try using YATE as a Google Voice Gateway.  It can be installed on either a separate server, or on the same server as your FreeSWITCH or Asterisk installation, however if you are running virtual machines then I recommend the separate server approach.  In fact, that may be the only way to do it with FreeSWITCH if you installed FreeSWITCH under Debian or Ubuntu, since the YATE install requires CentOS.  If you are a Linux expert you may be able to get around this, but don’t ask me how.

To install YATE, see this article from Nerd Vittles:

YATE in a Flash: Rolling Your Own SIP to Google Voice Gateway for Asterisk

EDIT: You may want to upgrade YATE to the latest version.

Just follow the instructions there, and the ones that you see after running the script to add a Google Voice user, and you should be fine, if you are using Asterisk.  The only things I would suggest that are not shown in those instructions are that you set your Trunk “Maximum Channels” to 2, because a Google Voice account will only permit two simultaneous channels of usage maximum, and that if YATE is on a separate server with a static IP address then I’d suggest adding permit/deny lines to the Asterisk Trunk PEER details to enhance security, like so:


Make sure the lines appear in that order, and replace xx.xx.xx.xx with the static IP address of the YATE server.  This may not help much because Asterisk is registering with the YATE server, but it can’t hurt either.

Also, you might want to consider changing the context statement to


to remove the +1 from the start of the Caller ID number on incoming calls.

The instructions don’t tell you to add a Dialed Number Manipulation Rule to your trunk configuration, but if you want to allow ten digit calls from any of your endpoints then you should add one rule that prepends 1 to 10 digit calls:

1+NXXNXXXXXX (The 1 goes in the first field, the NXXNXXXXXX in the third field)

If you are using the CallerID Superfecta module, and you use “Trunk Provided” as one of your data source, then after adding a Google Voice account to YATE I suggest editing /usr/local/etc/yate/regexroute.conf on the YATE server. You may need to install an editor first. For example, to install nano and then edit the file:

yum install nano
nano /usr/local/etc/yate/regexroute.conf

Look for the [contexts] section and there you will see a line for each of your Google Voice accounts that looks like this:


Just add ;callername to the end of each such line:


This will make sure that nothing is sent for a Caller ID name, so that Caller ID Superfecta will recognize that there is no “Trunk Provided” name and attempt to do a name lookup (note that you could also use ;callername=something to set the Caller ID name to a specific value). If you want to have ;callername
automatically appended whenever you create a new account, just use an editor to edit the script you use to add users, and find the line that looks like this (it should be near the bottom of the script):


Add ;callername to the end of the line, like so:


Save the modified file, and any time you add a new user it will automatically write that line with ;callername appended.

Thanks to Bill Simon for telling me about this method of sending the blank Caller ID name. Alternately, if you don’t want to mess with the YATE configuration, you could add a new Caller ID Scheme in Caller ID Superfecta that is only used with your Google Voice DID’s and that doesn’t include “Trunk Provided” as a data source.

Whether you are connecting from Asterisk or FreeSWITCH, if YATE is running on a separate server and the other system can’t register with YATE, it may be a firewall issue on the YATE server.  After I did the install I found that iptables was configured to only allow incoming ssh connections.  I modified that rule to only allow incoming ssh from a particular IP address (the one I’d be coming in from) and then added rules to permit traffic from the two servers allowed to talk to that YATE server.

EDIT: Hopefully this will not affect you if you have upgraded YATE to the latest version, but if you have a moderate number of Google Voice accounts, you may experience another issue.  If you start seeing messages like this when you telnet to YATE and then use debug on to see what is happening:

<sip:MILD> Flood detected: 20 handled events

And if every so often, the server appears to go into a semi-catatonic state, where calls come in but they don’t go out (this happened to me at least twice before I figured out what was happening), then you may have this issue.  It occurs when you have the same Asterisk server using multiple trunks to connect to YATE.  It turns out that whenever you reload Asterisk (as you might after making a configuration change, for example the “orange bar reload” in one particular GUI), it resends all of the registrations at once, and gives them all a default timeout of 120 seconds, so they all attempt to re-register at the exact same intervals.  And if you have several trunks, there are a LOT of SIP packets sent.  Plus, with qualifyfreq value set to 240, that means that every other time the registrations are taking place, qualifies are also taking place at the same time.  It appears that this is sufficient to cause that warning to appear once in a while.

The method I found that seems to fix this may not be the best way (feel free to comment if you know a better way), but it’s one way to deal with it.  What you need to do is change the registration expiration on each individual trunk so they are not all the same.  In Asterisk this can be accomplished by adding both of these settings to the trunk configuration (susbtitute nn with some random number of seconds, say between 90 and 120, and make it the same for both settings in each trunk, but different for different trunks)

In the trunk PEER details, add:


In the Register String, add  ~nn  to the end of the line, replacing nn with the same value used in the defaultexpiry setting, like so:

You might also need to vary the qualifyfreq value a bit in each trunk, so that it’s a bit under the specified 240 seconds and different for each trunk.  If doing those things doesn’t fix the issue, and you still get the <sip:MILD> Flood detected: 20 handled events message frequently, that could mean you are being subjected to an actual SIP attack.  The YATE installation includes a script with the filename /usr/src/yate/share/scripts/banbrutes.php that can be used to deal with that issue, but it’s not enabled by default.  View the banbrutes.php script in a text editor, and you’ll find instructions at the beginning of the script.  Or, you could tighten up the iptables firewall to only allow traffic from systems that are supposed to be talking to your YATE server.


As for FusionPBX, when you create a new Google Voice account on the YATE server using the provided add-yate-user script, at the end it will give you a bunch of configuration information for Asterisk.  These translate to FusionPBX Gateway settings as follows (showing what the script prints and the equivalent FusionPBX Gateway settings):

Trunk Name: YIAF1 ; or increment 1 if more than one (in FusionPBX I suggest you don’t use this; instead use the same setting as the Username for the Gateway name, particularly if you plan on having more than one Google Voice account)

host=x.x.x.x (Proxy in FusionPBX)
username=GV1234567890 (Username in FusionPBX)
secret=password (Password in FusionPBX)
type=peer (Not needed in FusionPBX)
port=5060 (Not needed in FusionPBX)
qualify=yes (Not needed in FusionPBX)
qualifyfreq=240 (Not needed in FusionPBX)
insecure=port,invite (Not needed in FusionPBX)
context=from-trunk (Not needed in FusionPBX)

Register String: … (Not needed in FusionPBX)

In FusionPBX, set Register to True and Enabled to True, and leave other Gateway settings at the defaults (EDIT: however, if you have several gateways to YATE, you might want to use the Expire seconds setting in FusionPBX to vary the registration timeouts a bit so that all your accounts aren’t trying to re-register at exactly the same time — see the longer EDIT section above for details).  Note that after you save the settings, it may take a few seconds for the state to change to REGED, so refresh the Gateways page after a bit and it should be okay if everything is configured properly and there are no firewall issues.

For your Inbound Route in FusionPBX, just use the Trunk Name/Username as the Destination Number (including the leading “GV“, which you can also use it in the Inbound Route name field if you like) and then choose the appropriate Action. When you first create the Inbound Route it will complain if you try to save a Destination Number that is not completely numeric, so just use any number and save the settings, then go back and edit the Destination Number field and also the Data field for the destination_number condition (which should be something like ^GV1234567890$, substituting your Google Voice number for the digits, of course).

For your Outbound Route, select your Google Voice trunk as the Gateway, and then select “11 digits long distance” from the dropdown in the “Dialplan Expression” setting. Save that, and if you only have one Google Voice trunk for all users on the system, that is all you need to do.  However, if you want to have multiple Google Voice trunks and have certain extensions only have access to certain trunks, the edit the Outbound Route you just created, and in the “Conditions and Actions” section at the bottom of the page, edit the last action on the page (the “bridge” action).  You want to change the Data field – it will contain something like sofia/gateway/GV1234567890/$1 and you want to change that to sofia/gateway/${accountcode}/$1 — save that change, and then when the Outbound Route page reappears, you may want to change the name to ${accountcode}.11d and add a Description like “Google Voice: Extension Account Code = Gateway Name” so you understand what the route is doing.  This single Outbound Route will handle all your Google Voice calls from all your extensions, if the Account Code setting for each Extension is set to the name of the Gateway for the Google Voice account you want that extension to use.

Note that if you are running PBX in a Flash, you can use the “Caller ID Superfecta” module to try to get a Caller ID name.  IF YATE itself has any ability to do Caller ID name lookups, someone will have to tell me how to enable and configure it, because at this point I would have no clue.  If you know, please leave a comment!

EDIT: To keep the YATE log file from growing too large over time, copy the file /usr/src/yate/packing/yate.logrotate into /etc/logrotate.d as “yate” (get rid of the .logrotate extension).  That file instructs the system logrotate job to roll the yate log file when it gets to 100 MB.  Thanks to Bill Simon for that tip!

EDIT 2: If you have ignored the advice given almost everywhere to create a new, separate Gmail account, and then use that account when you create your Google Voice account, then you have probably run into the issue of not receiving your incoming calls when you are logged into that Google account and for some time thereafter.  That problem, and one possible fix (along with the drawbacks) were discussed in a post in the thread “YATE in a Flash 1.2 Ready” on the PBX in a Flash Forum, which unfortunately disappeared from that site due to a server crash.  The post, originally by user Marian on Aug 6, 2012, read as follows:

Gmail sets a greater resource priority when you connect and don’t advertise unavailable for a while after you disconnect.
So, if you connect to GMail using the same account as yate the calls will be sent there until GMail advertise resource unavailable.
You can set priority=10 in accfile.conf, gmail account section.
But, if you do that you might not see your chat in GMail or another jabber client connected to GMail for the same account (like GTalk or Yate Client).
Unfortunately, the jabber protocol don’t allow setting different priorities for the same resource for different services (e.g. you can’t set a priority for chat and another one for another capatibility, like jingle calls).
I didn’t found a workaround for this situation: having, for the same account, a resource for chat and another one for jingle calls.
This would require a custom jabber client or a custom jabber server.

That, coupled with information from other posts around the web, means the best advice is to add a line of the form:


in each of your Google Voice accounts in the file accfile.conf (in the /usr/local/etc/yate directory).

If you want that line to be added by default when you add a new Google Voice account to your YATE server, open the add-yate-user script (which is probably in your /root directory) in a text editor such as nano, and find this line:

echo “options=allowplainauth” >> accfile.conf

and underneath it add this:

echo “priority=127″ >> accfile.conf

Then save the edited file.  I make no guarantees that this will actually work, but it’s worth a try. NOTE: The thread mentioned above suggested setting the priority to 10, however, the Asterisk developers are now using 25. As this wiki page explains:

More about Priorities

As many different connections to Google are possible simultaneously via different client mechanisms, it is important to understand the role of priorities in the routing of inbound calls. Proper usage of the priority setting can allow use of a Google account that is not otherwise entirely dedicated to voice services.

With priorities, the higher the setting value, the more any client using that value is preferred as a destination for inbound calls, in deference to any other client with a lower priority value. Known values of commonly used clients include the Gmail chat client, which maintains a priority of 20, and the Windows GTalk client, which uses a priority of 24. The maximum allowable value is 127. Thus, setting one’s priority option for the XMPP peer in res_xmpp.conf to a value higher than 24 will cause inbound calls to flow to Asterisk, even while one is logged into either Gmail or the Windows GTalk client.

Outbound calls are unaffected by the priority setting.

This would be true in Asterisk OR YATE, therefore the recommendation is to now use at least 25 as the priority value, up to the maximum of 127 as suggested above.

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


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:


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

# filename:
# A script to update iptable records for dynamic dns hosts.
# Written by: Dave Horner (
# 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

CHAIN=”dynamichosts” # change this to whatever chain you want.

# 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.”

# 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.”

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

if [ -a $HOSTFILE ]; then
# echo “CAT returned: $?”

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

# save off new ip.

echo “Updating $HOST in iptables.”
if [ “${#OLDIP}” != “0” ]; then
echo “Removing old rule ($OLDIP)”
echo “Inserting new rule ($IP)”

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,`

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 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 ( and and that works fairly well. Note that if ALL your Dynamic DNS addresses are from then you may want to change one line in the above script, from

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


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

This change will specify that the 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 or 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 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
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 (use http://… not https://…)

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

If you use choose to use instead of (which you might because 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:

Replace YOUR_DYNU_DYNAMIC_DNS with your dynamic DNS domain name, YOUR_DYNU_USERNAME with the username you use to log into your account, and MD5_HASH_OF_PASSWORD with the MD5 hash of your 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 and 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.

Link: Monit: Disk space monitoring


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!

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