Asterisk 10 (1.10) SMS (messaging or SIP Messaging) in action

[UPDATE: WORKS FLAWLESSLY WITH ASTERISK 11 TOO…On Asterisk 11, you can choose XMPP as well as the protocol, right now its just SIP:, imagine the possibilities. Also fixed the issue when failed message]

[ANOTHER UPDATE FOR Offline message - We all know when devices are offline, messages cannot be sent and it will be discarded by Asterisk. I've written another dialplan/solution to that here] 

With Asterisk 10 comes a channel independent dialplan for handling SIP MESSAGING (and jabber if that’s configured) method. In other words, phones that support SIP Messaging can send and receive “SMSes” through Asterisk 10 without the need to dial a channel making it like a regular telco SMS. Older Asterisks requires you to dial and answer a channel before you can send messages (see here).

It worked for me well in Asterisk 10.2.1 and this guide takes you through the process of using SMSing with a dialplan in a four step setup below. Asterisk uses the Message/ast_msg_queue channel to do all SIP Method MESSAGE related processing.

I was using FreePBX 2.10 but I guess it doesn’t quite matter as I am using a custom context anyway.

1) You need to modify your SIP general settings in sip.conf [freepbx users use the SIP Settings in Web GUI and add inside “Other SIP Settings”] add these two lines.

Non FreePBX users, edit sip.conf, under general
accept_outofcall_message=yes
outofcall_message_context=astsms

Save and exit.

We will now create the dialplan inside context astsms [FreePBX users edit the extensions_custom.conf file and put these in]

2) Non freepbx users, edit your extensions.conf file add the following
[astsms]
exten => _.,1,NoOp(SMS receiving dialplan invoked)
exten => _.,n,NoOp(To ${MESSAGE(to)})
exten => _.,n,NoOp(From ${MESSAGE(from)})
exten => _.,n,NoOp(Body ${MESSAGE(body)})
exten => _.,n,Set(ACTUALTO=${CUT(MESSAGE(to),@,1)})
exten => _.,n,MessageSend(${ACTUALTO},${MESSAGE(from)})
exten => _.,n,NoOp(Send status is ${MESSAGE_SEND_STATUS})
exten => _.,n,GotoIf($["${MESSAGE_SEND_STATUS}" != "SUCCESS"]?sendfailedmsg)
exten => _.,n,Hangup()
;
; Handle failed messaging
exten => _.,n(sendfailedmsg),Set(MESSAGE(body)="[${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)}] Your message to ${EXTEN} has failed. Retry later.")
exten => _.,n,Set(ME_1=${CUT(MESSAGE(from),<,2)})
exten => _.,n,Set(ACTUALFROM=${CUT(ME_1,@,1)})
exten => _.,n,MessageSend(${ACTUALFROM},ServiceCenter)
exten => _.,n,Hangup()
exten => _.,n,Hangup()

The above dialplan also send a little error message if the SMS didn’t send out saying its from “ServiceCenter”…

3) Save and exit and reload asterisk. [FreePBX users do #amportal restart]. Send SMSes! Yeah! But make sure you use a phone that supports SIP SMS.
IMPORTANT: In the SMS message center settings in your phone, enter the IP address of your Asterisk Server. In my case, I was using Unidata WiFi phone. As usual, do let us know the outcome and using other phones.
Here’s a message I got using the phone

Confirmed working with:
1) Unidata phones
2) Yealink phones (with SMS capabilities)
3) CSIPSimple on Andriod
4) X-Lite softphone
5) Most phones that support SIP Messaging

TechDay Asia - My Slides

TechDayAsia | TechDayAsia on Facebook During TechDay Asia 2012 (in Kuala Lumpur), i presented about Asterisk but more on a business pitch. The idea of the presentation is to let people know about how to start using Asterisk in various ways without changing much from the current setup you may have. Sort of like an integration.So, i gave a couple of scenarios and case studies from our years of requests by various customers. I wanted to share this to everyone out there who wants to get to know Asterisk and start using it without the hassle of a greenfield approach or complete makeover.TechDay-Asia - Sanjay

View more PowerPoint from sanjayws

And here's me presenting. And some pictures from TechDay which was a huge success. Spanning across three countries and cities.

Well done Astiostech boys!

Asterisk / FreePBX limit incoming calls based on DID (on ZAP/Dahdi/SIP) with FreePBX

Image Source: http://appcrawlr.com/android-apps/best-apps-restrict-access

So, we had this challenge by our customer to do this as they are using PRI and supporting multiple customers. Each customer needs to be limited to n number of channels on PRI. When they were using analog that was simply straightforward, its a physical line, so nothing much you can do about “limiting” it is limited by design!

The following guide allows you to limit calls based on

• A single DID
• A group of DIDs (in this guide the amount of DIDs per group is limited to 5, add more, improv as you wish)
• Group DIDs will be a union meaning, if you have DID1 and DID2 with limit of 3 calls, at any one time either calls coming to those DIDs are added up and if exceed 3, it will hangup.

Anyway, here’s a quick how-to to give you an idea how to go about it. Improv as you see fit :-)

Requirements: (my system)

1) FreePBX 2.10 or 2.11
2) Asterisk 1.8 or higher
3) Dahdi based PRI or SIP or just about anything with the use of proper declarations (variables)
4) Use MySQL
5) Debian Wheezy
6) Adminer to run a few MySQL tasks such as creating db/tables, editing values in them etc…

So here’s how:

1. Create a database inside MySQL called LIMITER
2. Use adminer and paste the following codes to using the “SQL Command” feature
   

   USE ` LIMITER`;
   CREATE TABLE `tbl_didlimiter` (
   `group` int(255) NOT NULL AUTO_INCREMENT,
   `data` varchar(100) DEFAULT NULL,
   PRIMARY KEY (`group`)
   ) ENGINE=MyISAM DEFAULT CHARSET=latin1;

3. Now, we will create a user superuser with password dbgod00, paste the following codes in SQL Command again
   

   CREATE USER 'superuser'@'localhost' IDENTIFIED BY ‘dbgod00';
   GRANT ALL PRIVILEGES ON LIMITER.* TO 'superuser'@'localhost';
   FLUSH PRIVILEGES;

4. Now, we edit the dialplan, the most important part! Since i use FreePBX, we use the _custom.conf to add new hacks. So that’s exactly what we are doing here, copy and paste the codes into extensions_custom.conf, like below
   
   #nano /etc/assterisk/extensions_custom.conf
   
   ;; DIALPLAN START ;;
   
   ;; READ ME FIRST
   ;; copyleft sanjay@astiostech.com
   ;; 1. Set trunks to use from-pstn, from-dahdi, from-zaptel OR from-trunk contexts
   ;; 2. The bold highlights may need to be changed depending on what you see in the
   ;; channel variables, most cases we see either EXTEN or FROM_DID or even CALLERID(DNID)
   ;; 3. Be sure this value is available and matching each other, in my case, the value
   ;; ${EXTEN} eventually matches the value ${CALLERID(DNID)} and they must
   ;; 4. This only supports one unique DID entry in DB. IF there are multiple entries
   ;; by mistake or whatever, it will pickout the first result that returns only
   ;;
   ;; 5. Feel free to add more G numbers as shown below, right now its just 5
   ;; 6. Maxdefault is set for global when no DB definitions are found for that DID and its limit,
   ;; if you do not want blanket settings, simply set MAXDEFAULT to blank or “”, if you want to set a global
   ;; limit then set it with MAXDEFAULT which then applies to all DIDs not set in DB.
   ;; Only when there’s a value found in DB then that DB value’s limits overrides maxdefault 
   
   [from-pstn-custom]
   exten => _X.,1,NoOp(Handling incoming to do cool stuff)
   same => n,Set(GROUP()=${EXTEN})
   same => n,Macro(didchoke)
   
   [macro-didchoke]
   exten => s,1,NoOp(Checking for incoming limits and applying if needed)
   exten => s,n,Set(MAXDEFAULT=””)
   exten => s,n,MYSQL(Connect connid localhost superuser dbgod00 LIMITER)
   exten => s,n,MYSQL(Query resultid ${connid} SELECT data from tbl_didlimiter where data like '%${CALLERID(DNID)}%' LIMIT 1)
   exten => s,n,MYSQL(Fetch fetchid ${resultid} DBRESULT)
   exten => s,n,MYSQL(Clear ${resultid})
   exten => s,n,MYSQL(Disconnect ${connid})
   exten => s,n,ExecIf($["${DBRESULT}"=""]?Set(DBRESULT=${CALLERID(DNID)})
   exten => s,n,GotoIf($["${DBRESULT}"=""]?exception)
   ;
   exten => s,n,Set(GROUPLIMIT=${CUT(DBRESULT,:,2)})
   exten => s,n,ExecIf($["${GROUPLIMIT}"=""]?Set(GROUPLIMIT=${MAXDEFAULT})
   exten => s,n,GotoIf($["${GROUPLIMIT}"=""]?exception)
   ;
   exten => s,n,Set(DIDS=${CUT(DBRESULT,:,1)})
   exten => s,n,Set(DID1=${CUT(DIDS,\,,1)})
   exten => s,n,Set(DID2=${CUT(DIDS,\,,2)})
   exten => s,n,Set(DID3=${CUT(DIDS,\,,3)})
   exten => s,n,Set(DID4=${CUT(DIDS,\,,4)})
   exten => s,n,Set(DID5=${CUT(DIDS,\,,5)})
   ;
   exten => s,n,ExecIf($["${DID1}"!=""]?Set(G1=${GROUP_COUNT(${DID1})}))
   exten => s,n,ExecIf($["${DID2}"!=""]?Set(G2=${GROUP_COUNT(${DID2})}))
   exten => s,n,ExecIf($["${DID3}"!=""]?Set(G3=${GROUP_COUNT(${DID3})}))
   exten => s,n,ExecIf($["${DID4}"!=""]?Set(G4=${GROUP_COUNT(${DID4})}))
   exten => s,n,ExecIf($["${DID5}"!=""]?Set(G5=${GROUP_COUNT(${DID5})}))
   ;
   exten => s,n,ExecIf($["${DID1}"=""]?Set(G1=0)
   exten => s,n,ExecIf($["${DID2}"=""]?Set(G2=0)
   exten => s,n,ExecIf($["${DID3}"=""]?Set(G3=0)
   exten => s,n,ExecIf($["${DID4}"=""]?Set(G4=0)
   exten => s,n,ExecIf($["${DID5}"=""]?Set(G5=0)
   ;
   exten => s,n,Set(TOTALGGROUPCHANS=$[${G1}+${G2}+${G3}+${G4}+${G5}])
   exten => s,n,NoOp(So total channels here are ${TOTALGGROUPCHANS} of GROUPLIMIT of ${GROUPLIMIT})
   exten => s,n,GotoIf($["${TOTALGGROUPCHANS}" > "${GROUPLIMIT}"]?overlimit)
   exten => s,n,MacroExit()
   ;
   exten => s,n(overlimit),Busy(20)
   exten => s,n,Hangup(16)
   exten => s,n,MacroExit()
   ;
   exten => s,n(exception),MacroExit()
   
   ;; DIALPLAN END ;;

5. Now, reload asterisk dialplan, be sure to tail the log file to start troubleshooting if things don’t go right.
   #asterisk -rx “dialplan reload”
6. Now, edit the DB values and add DIDs like show in example below, use adminer or similar for easy editing
   E.g. 1 Format: DID1:3
   Where DID1 is DID you wish to limit to 3 channels
   E.g. 2 Format: DID1,DID2,DID3,DID4,DID5:3
   Where DID1-5 are the DIDs you wish to limit to 3 channels combined
   
   Here’s sample data from my own server!
      

Fire away, test it out…! As usual, appreciate the feedback and ideas to improve! Do let us know how it went for you!

How to explain Nagios to grandma

So I visited grandma couple of days back and while talking about work and politics and all, she asked, so what’s this Nagios thing you guys are experts at?
I replied, its an “Enterprise Monitoring Solution” and she went like “Enterprise what now….”, and that actually got me thinking deep (something that i rarely do).
So anyway, to answer gramps pressing question, I gave this analogy.

Take for example a car


One thing cars these days come with are essential gauges to let us know what’s going on inside this technically complex electrical and mechanical marvel. This is called a dashboard.
So dashboards in a car gives us all these really cool information; and while cool, they’re also very essential for the well being and running of that car.

Most modern cars, you can tell the fuel levels, battery status, gear status, hand breaks statuses and what not. All essential right? And some more expensive cars like beamers, will even tell you that you’re due for a service soon!. Now, like many of us, the first thing I do when I power up my baby is to check this dashboard and see all that’s going on and when things are fine, engage gear, pull down that windscreen and breathe all the lovely smog from Kuala Lumpur air. {deep chesty cough x 2}
Now here’s where it gets difficult, lets take the challenge of watching over 10, or 100 or even 1000 cars all at the same time and to know at point in time and accuracy if one of these babies of yours isn’t behaving properly. While its possible to do one at a time but sometimes, it may be too late before you reach car no 99 to know that the battery has been completely discharged. Ouch!..can’t use that car until it gets fixed now… :-(

Fear not car lover, here’s where Nagios comes to play. It is designed in such a way, doesn’t matter which type of car you drive and how the dashboard works, if it is in that dashboard, we can take all the important status information and put them in an easy to use web page for all those cars you manage.
Not only that, when a car reaches a fuel level of less than 10% (or anything you think its important to know about before it goes bonkers), Nagios can..
a) email you
b) send you smses
c) call you
d) put up a message on facebook for you
e) set an appointment with you mechanic
And there ya go, she got it, and thereafter, she went back straight to her room to sleep. But I knew she understood, I do….
Now back in geek-land…
Imagine those cars as servers, devices, applications/software, virtual or cloud infrastructure and what not. Each of these in some small or elaborate way have their own “dashboards” that come in a form of a simple log entry to a popups on your device saying something is wrong and what not. Again, to manage just this one, perhaps its very fine. But imagine managing hundreds, on top of that, a single server can potentially host tens of important apps, making it alone “a lot” to monitor.
The cool thing about Nagios is, you can really literally monitor whatever that server/apps/device  shows you, no gimmicks no joke..Monitor just about any type of hardware, OS, application and of a bare minimum, a plain ole log file!
Give it a try and see the power of Nagios for yourself.
Happy weekend.

Nagios Plugin–Advance Traceroute to check between two devices

We had to create a plugin to basically do the following
1) Do a typical traceroute from the Nagios box to a destination IP
2) Instead of calculating the time between the Nagios to Destination Host, we are interested to know the time between two host in between

In other words, a typical traceroute will
NagionServer –> Gateway –> Hop 1 –> Hop 2 –> Hop 3 –> Destination

What this plugin can do is when defined correctly, to check the time (in ms) between Hop 1 up until Hop 3, plot a graph and put up warning and critical values for your alerting.
Here’s the sample plugin, and relevant configuration files you probably need.
NOTE: You may need to tweak for different  Oses other than Debian as this was created and tested with a Debian.
The plugin

• The plugin (place typically in /usr/local/nagios/libexec)
• Paste below into a file say trace_time
• Make sure it belongs to user <nagios> and has execution right; e.g.
• chown nagios:nagios /usr/local/nagios/libexec/trace_time
• chmod +X /usr/local/nagios/libexec/trace_time

#####START PLUGIN##### #!/bin/bash # # usage # ./trace-time <final-dest> <startip> <endip> <warning> <critical> # Note: You must define all three, there's no error checking # tip: do a traceroute first, then determine from which ip to which ip do you want to calculate. If # # DEST=$1 IP1=$2 IP2=$3 WARNING=$4 CRITICAL=$5 PROG=`which traceroute` if [[ $DEST == "" ]]; then         echo "UNKNOWN: No destination ip defined"     exit 3 fi if [[ $IP1 == "" ]]; then         echo "UNKNOWN: No start ip defined"     exit 3 fi if [[ $IP2 == "" ]]; then     IP2=$DEST fi if [[ $WARNING -eq "" ]]; then         echo "UNKNOWN: No warning value defined"         exit 3 fi if [[ $CRITICAL == "" ]]; then         echo "UNKNOWN: No critical value defined"         exit 3 fi if [[ $WARNING  >  $CRITICAL ]]; then         echo "UNKNOWN: Warning value larger than critical value"         exit 3 fi # myepoch=`date +%s` filename=/tmp/$myepoch.tmp.txt tempfile=/tmp/$myepoch.output # /bin/touch $filename /bin/touch $tempfile # /bin/chown nagios:nagios $filename /bin/chown nagios:nagios $tempfile # # getreading=`$PROG -n -q 1 $DEST > $tempfile` # numberip1=`cat $tempfile | grep ms | grep $IP1 | awk {'print $1'}` numberip2=`cat $tempfile | grep ms | grep $IP2 | awk {'print $1'}` # # for i in $(seq $numberip1 $numberip2) do         getms=`cat $tempfile | sed -e 's/^[ \t]*//' | grep ^$i |  awk {'print $3'}`     echo $getms >> $filename done # startcalc=`awk '{s+=$0} END {print s}' $filename` # rm $filename rm $tempfile # # OUTPUTS # grapher="$IP1-->$IP2" # if awk 'BEGIN{if(0+'$startcalc'>'$CRITICAL'+0)exit 0;exit 1}' then         echo "CRITICAL($startcalc): Time exceed critical value|$grapher=$startcalc;$WARNING;$CRITICAL"         exit 2 fi if awk 'BEGIN{if(0+'$startcalc'>'$WARNING'+0)exit 0;exit 1}' then         echo "WARNING($startcalc): Time exceed warning value|$grapher=$startcalc;$WARNING;$CRITICAL"         exit 1         else             echo "OK($startcalc): Time OK|'$grapher'=$startcalc;$WARNING;$CRITICAL;;"         exit 0 fi #####END PLUGIN#####

Nagios – Host.cfg

define host{         use                     debian5-linuxserver         host_name     Google WWW server         alias                   For Tracing TimeHop Distances         address            209.85.175.105         }

Nagios – commands.cfg

define command{         command_name    check_time_between_hosts         command_line    $USER1$/trace-time $HOSTADDRESS$ $ARG1$ $ARG2$ $ARG3$ $ARG4$          }

Nagios – services.cfg

define service{         use                                       debian5-linuxservice         host_name                       Google WWW server         service_description      Between IP 210.5.40.153 to 209.85.250.237         action_url                          /nagios/pnp/index.php?host=$HOSTNAME$&srv=$SERVICEDESC$         check_command            check_time_between_hosts!210.5.40.153!113.23.161.66!10!20        }

* Note, the template debian5-linuxservice and debian5-linuxserver is not default and you need to define one first or use the defaults
Now, just restart Nagios to make it work.

More info
In order for you to know the hop you wish to monitor, simply do a traceroute;
# traceroute -n -q 1 209.85.175.105
-n = Numeric output
- q 1= Only do a single query
In this example below, I am tracing to one of Google’s servers at 209.85.175.105, the output of the trace is like below (NOTE!: actual IPs have been changed)

1  111.22.42.3  0.554 ms
2  111.22.40.153  0.667 ms
3  111.22.40.125  1.026 ms
4  203.188.233.121  1.218 ms
5  203.188.233.205  1.488 ms
6  113.23.161.66  1.627 ms
7  209.85.242.246  1.542 ms
8  209.85.242.125  2.322 ms
9  66.249.94.158  3.075 ms
10  209.85.175.105  2.801 ms
So lets say you wish to trace the time between IP 111.22.40.153 and IP113.23.161.66, simply use the plugin with these values on the CLI (to test);
# ./trace-time 209.85.175.105 111.2.40.153 113.23.161.66 10 20
And the output will look like this;
OK(5.909): Time OK|'111.22.40.153-->113.23.161.66'=5.909;10;20;;
*Which is a typical output expected by Nagios with PNP graphing enabled
Graphs will look like this

pfSense and TM Unifi

This is to show you how to enable pfSense 2.x and use TMNet’s Unifi.
Important to note that inorder to “dial” into Unifi, you must fire the PPPoE through a VLAN and those are VLAN 500 for your regular internet browsing and VLAN 600 for your Video. This guide is to setup basic internet using pfSense.
Assuming you are using at least 2 interfaces, the LAN (in this guide, eth0), which all your local networks (or regular users) are connected, the other interface is WAN (in this guide, eth 1) which will be the one that will dial into Unifi.


Your LAN can be easily configured to any IP which you wish your users to connect to. This will typically be the gateway IP address for your users. The interface we will discuss here is the WAN interface.
So summary of the steps to get it working;
1) Define the WAN interface as null (or none)
2) Setup VLAN 500 and tie it to the WAN interface or eth1
3) Setup PPPoE inside VLAN interface

1) Defining WAN interface as null.
Simply go to WAN interface and set it up as none. Save it. Setup like below


2) Define a new interface:
On pfSense’s main menu, click on Interfaces and click on (assign). Click on VLANs. Select WAN (or eth1) as the parent interface. Inside VLAN tag, enter 500. Put a nice description, like VLAN-TMUnifi there if you so wish. Click on save.

3) Setup PPPoE on the VLANned interface:
Now, click on Interfaces again, click on (assign). Click on the + to add a new interface and select the newly created VLAN-TMUnifi. Notice the interface name that was created for you like OPT1 or OPT2. Click on interfaces again, select (and in this example), OPT1. Click to enable the interface. Under type, select PPPoE. Scroll down to the username and password part. Enter your username and password respectively.
You can also rename your interface as for example, from OPT1 to UNIFPPP (for example):


Before you click on save, make sure you select, Block private and bogon networks. They should not pass through this interface (unlike interface LAN).
Now, once you’ve save, click on the blue S icon on the top right of the page. If all config such as username and password is correct, you should see the interface come up like below;

If you have problems, click on, Status, click on System Log files then click on PPP. Any issues or errors will be shown there.

Once done, you need to setup gateways and routing correctly in order to route traffic over this new interface you just created and dialed.

You should set it up as default unless you have other gateways and such. Otherwise, setting it as default will quickly allow you to browse.
Well that’s it, also of course don’t forget to setup an allow rule from interface LAN to any (to begin with) and allow that connection.
If you need to allow incoming connections, select the UNIFPPP interface instead of WAN and setup your inbound rules there.
Enjoy pfSense and the speed of Unifi!
PS> If you are interested in our appliance, do write to sanjay@astiostech.com. Should cost you around US250/box with 4G flash, enclosure, 3 NICs, 2 USB ports (can use for 3G usb modems etc).

Using Asterisk call files with FreePBX (using Local channel)

Call files are perhaps one of the coolest things you can do with Asterisk. Just literally dump the file in a particular directory and voila, you can make a call. Past few days, I’ve been playing around with it quite a bit and would like to share some things with you on using the Local channel which is most beneficial since you may wish to take advantage of the dialplans and call flows defined in FreePBX (for instance).
Now, the Local channel does a lot such as checks if voicemail is enabled, call forwarding (follow-me) and other customization to the user’s account. It also would support the Device and User mode if that’s being used. The Local channel rather than using technology channels directly can help with several things again for example restrictions that may apply (context) for a particular user. Using channels like SIP/1000 and IAX/1000 will literally bypass all the good stuff that may have been setup.
In conjunction with asterisk call files e.g. .call. You can do lots! including lots of automation and what not just like how you would do using AMI or any AGI stuff if you know about them. Here I would like to show you how to take advantage of the Local channel in call files.
Channel: Local/2000@from-internal/n
CallerID: <0386892800>
MaxRetries: 0
RetryTime: 1
WaitTime: 10
Context: from-internal
Extension: 86892888
Priority: 1 
Archive: no
Set: AMPUSER=2000
Now try modifying these above values to match your specific needs, e.g. channel 2000 is a user 2000 attached to device 1000, change that, change Extension to the destination you wish to bridge a call and here’s what’s important for device/user mode, you must specify the AMPUSER value otherwise, the macro-set-callerid will not be able to set right variables such as outbound CID for you since the Local/2000 will eventually be using SIP/1000 and SIP 1000 doesn’t really have anything more than just a connection agreement (username/password).
Create this file anywhere and then dump it into /var/spool/asterisk/outgoing and voila, you’ve got call.
Also, would like to share a script to automatically create and move files for you; note this will work with FreePBX 2.8 or higher and Asterisk 1.8.
1) nano /root/callgenerator.sh
2) Paste the script below

#!/bin/bash
user=$1
dial=$2

if [[ "$user" -eq "" ]]; then
    echo "ERROR No User / Destination Defined"
    exit 2
fi

if [[ "$dial" -eq "" ]]; then
    echo "ERROR No Destination Number Specified"
    exit 2
fi

# generate call file
mydate=`date +%d%m%y_%H%M%S`
filename="$mydate.U-$user-D-$dial.call"
#
# SUPPORT ONLY SIP
echo -e "Channel: Local/$dial@from-internal/n
CallerID:
MaxRetries: 0
RetryTime: 1
WaitTime: 10
Context: from-internal
Extension:*60
Priority: 1
Archive: no
Set: AMPUSER=$user" > /var/spool/asterisk/$filename

# Call now
# File ops
chown asterisk:asterisk /var/spool/asterisk/$filename
mv /var/spool/asterisk/$filename /var/spool/asterisk/outgoing/ > /dev/null 2>&1
if [[ $? -ne 0 ]]; then
    echo "ERROR Parsefile crashed"
    exit 2
else
    echo Start calling..
fi

exit 0

3) Make it executable chmod +x /root/callgenerator.sh
Test it out like this /root/callgenerator.sh 2000 986892888 (where 2000 is the user’s phone that will first ring and once picked up, it will proceed with the other leg of the call i.e. 986892888, its important here to use from-internal otherwise, your dialplan routing and prefix handingling (in this case, the number 9) will not be stripped off when calling).

FreePBX IVR direct dial across multiple locations/asterisk servers/FreePBX servers

Assuming you have 2 FreePBX servers across two location that are connected via a trunk and the trunk dialling does work fine. Meaning you can already call SiteA and SiteB and vice-versa.
SiteA: Extensions 1000 and on
SiteB: Extensions 2000 and on
With the default IVR context in FreePBX it’s limited to local numbers only (in this case SiteA extensions). Imagine if you had a PRI line coming into Site A and the IVR direct dial would then need to reach extensions on Site B, this will by default will not work.
Here’s how to “extend” the IVR dialling to other locations. NOTE, we assume the outbound route doesn’t have a prefix to dial intersite between Site A or Site B.
1) nano /etc/asterisk/extensions_custom.conf
2) Enter the following codes (here’s Site A IVR trying to reach site B)
[from-did-direct-ivr-custom]
exten => _2XXX,1,ExecIf($["${BLKVM_OVERRIDE}" != ""]?Noop(Deleting: ${BLKVM_OVERRIDE}: ${DB_DELETE(${BLKVM_OVERRIDE})}))
exten => _2XXX,n,Set(__NODEST=)
exten => _2XXX,n,Goto(from-internal,${EXTEN},1)
#NOTE each line should start with exten like below, on single lines

3) Save the file and reload the dialplan
asterisk –rx “dialplan reload”
Lets say you use prefix 7 to dial site B from site A followed by the extension number. Just edit the part and as an example below;
exten => _2XXX,n,Goto(from-internal,7${EXTEN},1)
Where 7 prefixes automatically is entered before the extension number.
Have fun!

The 4th Cloud Service? – VoIP As a Service (VaaS)

This weekend, i had a chance to present to Microsoft guests at their office here in Malaysia about cloud computing and focusing my talk on cloud technology security.
It was clear after a while jiving with those techies and seeing their responses and also doing much research, this concept, the relevant technologies wrapped around and more importantly, enjoying unwavering support from some of the big boys such as Google, Microsoft, Amazon, EMC, etc, one can easily deduce that the {{CLOUD}} is here to stay. This could mean a lot to consumers or business opportunists alike to ride on the wave (or as some may still insist, the hype)
In the weekend while on the treadmill at the local gym i started think about one particular service and of course being Astiostech’s core of 2 cores, Asterisk and VoIP and how it fits the 3 domains defined by the Cloud Security Alliance (CSA), and these are;
1) Software as a service
2) Platform as a service
3) Infrastructure as a service
Thinking about where would VoIP sit, it becomes quite clear to me after while that it doesn’t really belong to those categories (as far as I've understood them.) One would think it’s close to SaaS or Software as a service, but i beg to differ, Why?
1) VoIP is software in code, true, but its not offered as one, but rather a communication platform
2) Hosting VoIP is not the same as software, it breaks the boundaries of typical software running on computers and basically runs on IP phones rather.
3) It can connect to other software services to introduce Computer Telephony Integration or CTI to enrich one’s telephony experience but it in itself, isn’t one.
Hosted PBX isn’t something new. Astiostechtech has done so for almost 4 years now and rebranded it as e-fon ( www.efon.my ). We find such application of technology to be quite exciting then and over the years, seeing almost no downtime or complains, we are now convinced that it’s a technology and offering that could really change the way we do PBXes..
How has PBXes works for you currently and in the past.

Typically it was like the above diagram
1) Everything was at premise
2) Phones and servers communicated locally
3) Expansions were done as long as the PBX supports them
4) Most of the time, companies change their PBX when moving offices
5) Upgrades get done very sparingly as it can really blow a huge hole in the pocket
Lets talk about web servers and company websites, I can imagine almost all small and medium sized companies host their websites with local providers and do not host them internally. The reason why they host their websites is quite obvious, it’s because the service and support was there and it was just paying a very affordable monthly fee. This is closely tied to the economics of scale. They run operations of hundreds of servers and buy and rent stuff in bulk. Imagine a Hosted PBX like your Website server.
This elevates the need for running local servers, hiring experts to setup and run it and not to forget the cost of running and maintaining such infrastructure. Also to think about disaster recovery and business continuity, while, we businesses should be thinking about making money obviously, and not maintaining PBXes like we didn’t maintain our websites.
Now, imagine if the same like websites, were to be offered as a service to you in regards of telecommunications or simply, a hosted PBX instead of an on premise one.

Here’s a sample provider, like Astiostech (e-fon.my) can provide you. As an end user in a small and medium sized organization running your own onsite PBX is shortly becoming a thing of the past. Imagine again these infrastructures are now placed in cloud computing technology.
First and foremost; hosting outside your premise and into services like e-fon can help
1) Re-size on the fly
2) Gain mobility and do not lose your "favourite” phone numbers
3) It has all you need or even more than you can expect. (Economies of scale)
4) Its cheaper to have enterprise features, remember, the economies of scale! Make your organization sound like a multi-billion dollar setup (perception helps btw)
5) Connect to multiple types of Telcos that the PBX provider subscribes to and enjoy rates they enjoy as well
6) Connect to your other offices flawlessly if they use common protocols such as SIP or H323
7) Do not worry about upgrades as its done centrally
8) Use different types of phones or even free ones like softphones
Those you get with Hosted PBXes, i am sure or at least with e-fon you do. But here’s the kicker, with now cloud computing, e-fon gets hosted on cloud services. Not only you enjoy those highlighted list of benefits above, you also can enjoy that offered by putting stuff on cloud computing architecture such as;
1) Fabric on demand (CPU, disk, memory)
2) Auto and transparent redundancy
3) Strong network architecture and bandwidth
4) Probably more secure than if you ever had to put on premise
5) Ride on credible cloud providers for credible service offering
I can imagine in the next few years, this would be the “talk of town” per-se. Hosted PBX will be on just about all companies and hosting on premise even with a keyphone system isn’t attractive or make any sense anymore to business people. Just like how hosting websites took a while before just about all friends and companies i know in the small and medium category of companies have their websites with a provider or in a cloud alike.
From the Astiostechtech Engineering Office…

How to make FreePBX updates through Proxy

SInce FreePBX does not honour the proxy settings via the bash export, you need to tell Apache instead to use a proxy. This would apply for FreePBX as well.

Here’s a how to for Debians.

1) #nano  /etc/apache2/envvars

2) Add the following codes at the end. You will notice that there's repeats, sometimes they are needed so keep them there twice.

export ftp_proxy=http://10.50.19.206:8080
export http_proxy=http://10.50.19.206:8080
export FTP_PROXY=http://10.50.19.206:8080
export HTTP_PROXY=http://10.50.19.206:8080

3) #/etc/init.d/apache2 restart

Where 10.50.19.206 is the proxy server and 8080 is the proxy port, you can also insert an auth username like this for all instances above.

http://username:password@10.50.19.206:8080