PowerDNS Recursor – Hiding/Private Zones via LUA

When deploying PowerDNS Recursor for our customers, we had an issue with how to hide some responses/zones to external IPs. The way we implemented this solution is via LUA scripting.

-- Set Private Zone
myDomain = newDN("private.com")

-- Whitelist IPs
myNetblock = newNMG()

function preresolve(dq)
	if dq.qname:isPartOf(myDomain) and myNetblock:match(dq.remoteaddr) and dq.qtype == pdns.A then
		-- if the dns requests is for the private zone by trusted IP, proceed to respond.
		return false
	elseif dq.qname:isPartOf(myDomain) and dq.qtype == pdns.A then
		-- if the dns requests is for the private zone by UNtrusted IP, do not respond for A record requests
		pdnslog ("Remote IP " .. dq.remoteaddr:toString() .. " for domain=" .. dq.qname:toString() .. ", type=" .. tostring(dq.qtype) )
		return true
		-- function normally for everything else.
		return false

In the above example, we set the zone/domain that needs to be private, next select which IP addresses are allowed to see the records and finally the logic via the preresolve() function.

[FreePBX/Asterisk]Multiple / Rotary Call Flow Toggles

We got a customer the other day who wanted to manually override their normal call flow. This could have been easily achieved by a single toggle if it was to a single destination. But this customer wanted wanted to be able select multiple destination and also only wanted to push one button to do so.

This meant we had to write custom code to get this done and it was achieved by writing a code to manipulate call flow toggles.

What you need to add is the following to the extensions_custom.conf

include => enable-switch

exten => s,1,NoOp("This context loops through all the switches and checks which one is ON. Than it follows its destination")
exten => s,n,GoSub(sub-get-nr-of-switches,s,1)
exten => s,n,Set(nrofs=${GOSUB_RETVAL})
exten => s,n,Set(i=0)
exten => s,n,Set(default-dest=app-daynight,4,1)
exten => s,n,Set(switchnr="NULL")
exten => s,n,While($[$[${i} < ${nrofs}] & $[${switchnr} = "NULL"]])
exten => s,n,Set(switchnr=${IF($["${DB(DAYNIGHT/C${i})}" = "NIGHT"]?${i}:"NULL")})
exten => s,n,Set(i=$[${i} + 1])
exten => s,n,EndWhile
exten => s,n,GotoIf($[${switchnr}!="NULL"]?app-daynight,${switchnr},1:${default-dest})
exten => s,n,Hangup()

exten => s,1,Set(nr_of_s=0)
exten => s,n,Set(exten_state="NOT_ACQUIRED")
exten => s,n,While($[${exten_state}!=0])
exten => s,n,Set(exten_state=$[${VALID_EXTEN(app-daynight,${nr_of_s},1)}])
exten => s,n,NoOp(Exten nr ${nr_of_s} of app-daynight is ${exten_state})
exten => s,n,Set(nr_of_s=${IF($[${exten_state}=1]?$[${nr_of_s}+1]:${nr_of_s})})
exten => s,n,EndWhile
exten => s,n,NoOp(${nr_of_s} switches found)
exten => s,n,Return(${nr_of_s})

exten => _*20X,1,NoOp("Enable selected Switch and disable others")
same => n,GoSub(sub-get-nr-of-switches,s,1)
same => n,Set(nrofs=${GOSUB_RETVAL})
same => n,Set(switchtoactivate=${EXTEN:3})
same => n,Set(CSTATE=${DB(DAYNIGHT/C${switchtoactivate})})
same => n,NoOp(Activated Switch nr ${switchtoactivate})
same => n,Set(i=0)
same => n,While($[${i}<${nrofs}])
same => n,GotoIf($["${CSTATE}" = "NIGHT"]?end1)
same => n,Set(DB(DAYNIGHT/C${i})=DAY)
same => n(end1),NoOp(Dialing the same toggle code. Switch it off)
same => n,Set(DEVICE_STATE(Custom:AHTGL${i})=NOT_INUSE)
same => n,Set(DEVICE_STATE(Custom:DAYNIGHT${i})=NOT_INUSE)
same => n,Set(i=$[${i}+1])
same => n,EndWhile
same => n,Set(freepbx_toggle_dest=*28${switchtoactivate})
same => n,GotoIf($["${CSTATE}" = "NIGHT"]?end2)
same => n,Set(DEVICE_STATE(Custom:AHTGL${switchtoactivate})=INUSE)
same => n(end2),NoOp(Skip setting BLF)
same => n,Goto(app-daynight-toggle,${freepbx_toggle_dest},1)
same => n,Hangup
same => hint,Custom:AHTGL${EXTEN:3}

Then from FreePBX add custom destination with check-active-switch,s,1 as the value and point the inbound route to this custom destination.

Now, you create all the call flow toggles with their override mode set to the destinations you want to go and the normal mode is set identical on each toggle the normal operation you want to work.

Finally, create BLF keys to see which switch is active using *20X as the value to dial/monitor.

Modify the following line to point calls to a default destination if none of the toggles are on overrideĀ  mode. On the below code it is set to app-daynight,4,1

exten => s,n,Set(default-dest=app-daynight,4,1)

The original code is via : https://community.freepbx.org/t/multiple-daynight-toggles-aka-call-flow-control/27518

Modifications on this code helps you use BLF to monitor currently selected toggles as well as ability to turn off the the override.

Solving NAT related issues for Hosted PABXes

If you come across a situation where a customer has multiple handsets usually when upwards of 20 units per site/location, you may have noticed that you get myriad of issues depending on the router/firewall being used.

Some issues that I’ve noticed are, phones loosing registration randomly and unable to register again; Transfer/hold/BLF functions stops working correctly.

The main culprit here is that the router/firewall is unable to cope with NATing table for same source port for this many number of devices and sometimes freezes. The side affect of this is that traffic from the hosted PABX does not reach the handset as the router stops forwarding the traffic inwards.

There are two remedies that I have worked out. First being setting up a VPN, which opens a whole heap of other issues plus brings added complexity. The second being changing the listening port of the device to be unique one. This is much simpler and easier to maintain and troubleshoot.

Now the hard part is that if you have a site with hundreds of provisioning files for each phone, this would be tedious and cumbersome. So here’s a simple script to help you save the day. This is designed for Yealink phones and can be adapted to any other phone model.


#check old yealink phones are there
if ls 001565* 1> /dev/null 2>&1; then

while [ $LOOP -lt 2 ]; do
  for f in $FILES
    #check if sip.listen_port already exists. if not proceed to add following
    if ! grep -q '^[[:blank:]]*sip\.listen_port' "$f"; then				
      UNAME=$(awk '/user_name/{print $NF}' "$f")
      # take action on each file. $f store current file name
      # check the lenght of extension number and set the last 3 digits as listen port
      if [ ${#UNAME} -lt 2 ]; then
        echo "sip.listen_port = 900$UNAME" >> $f
      elif [ ${#UNAME} -lt 3 ]; then
        echo "sip.listen_port = 90$UNAME" >> $f
      elif [ ${#UNAME} -lt 4 ]; then
        echo "sip.listen_port = 9$UNAME" >> $f
        echo "sip.listen_port = 9$UNAME" >> $f
  #check new yealink phones are there
  if ls 805ec0* 1> /dev/null 2>&1; then


[Asterisk/FreePBX]Set Inbound DID as Caller-ID for Forwarded Calls

This is a snippet developed by a colleague of mine. This is for people who’s supplier’s doesn’t allow passing original Caller ID for forwarded calls. What this does is it will show the DID where the calling party called in as the caller ID instead of showing up your default trunk number for the caller ID.

First add the below code to extensions_override_freepbx.conf, next restart/reload asterisk and you’re good to go.

include => macro-outbound-callerid-custom
exten => s,1,ExecIf($["${CHANNEL(channeltype)}" = "Local"]?Set(CALLERID(all)=${MASTER_CHANNEL(DEXTEN)}))
exten => s,n,ExecIf($["${CHANNEL(channeltype)}" = "Local"]?Set(AMPUSER=${MASTER_CHANNEL(DEXTEN)}))
exten => s,n,ExecIf($["${CALLINGNAMEPRES_SV}" != ""]?Set(CALLERPRES(name-pres)=${CALLINGNAMEPRES_SV}))
exten => s,n,ExecIf($["${CALLINGNUMPRES_SV}" != ""]?Set(CALLERPRES(num-pres)=${CALLINGNUMPRES_SV}))
exten => s,n,ExecIf($["${REALCALLERIDNUM:1:2}" = ""]?Set(REALCALLERIDNUM=${CALLERID(number)}))
exten => s,n(start),GotoIf($[ $["${REALCALLERIDNUM}" = ""] | $["${KEEPCID}" != "TRUE"] | $["${OUTKEEPCID_${ARG1}}" = "on"] ]?normcid)
exten => s,n,GotoIf($["foo${DB(AMPUSER/${REALCALLERIDNUM}/device)}" = "foo"]?bypass)
exten => s,n(normcid),Set(USEROUTCID=${DB(AMPUSER/${AMPUSER}/outboundcid)})
exten => s,n(bypass),Set(EMERGENCYCID=${DB(DEVICE/${REALCALLERIDNUM}/emergency_cid)})
exten => s,n,Set(TRUNKOUTCID=${OUTCID_${ARG1}})
exten => s,n,GotoIf($["${EMERGENCYROUTE:1:2}" = "" | "${EMERGENCYCID:1:2}" = ""]?trunkcid)
exten => s,n,Set(CALLERID(all)=${EMERGENCYCID})
exten => s,n,Set(CDR(outbound_cnum)=${CALLERID(num)})
exten => s,n,Set(CDR(outbound_cnam)=${CALLERID(name)})
exten => s,n(exit),MacroExit()
exten => s,n(trunkcid),ExecIf($[${LEN(${TRUNKOUTCID})} != 0]?Set(CALLERID(all)=${TRUNKOUTCID}))
exten => s,n(usercid),ExecIf($[${LEN(${USEROUTCID})} != 0]?Set(CALLERID(all)=${USEROUTCID}))
exten => s,n,ExecIf($[${LEN(${TRUNKCIDOVERRIDE})} != 0 | ${LEN(${FORCEDOUTCID_${ARG1}})} != 0]?Set(CALLERID(all)=${IF($[${LEN(${FORCEDOUTCID_${ARG1}})}=0]?${TRUNKCIDOVERRIDE}:${FORCEDOUTCID_${ARG1}})}))
exten => s,n(hidecid),ExecIf($["${CALLERID(name)}"="hidden"]?Set(CALLERPRES(name-pres)=prohib_passed_screen))
exten => s,n,ExecIf($["${CALLERID(name)}"="hidden"]?Set(CALLERPRES(num-pres)=prohib_passed_screen))
exten => s,n,Set(CDR(outbound_cnum)=${CALLERID(num)})
exten => s,n,Set(CDR(outbound_cnam)=${CALLERID(name)})


Connecting two Asterisk/FreePBX using SIP Trunks

This was a project that I’ve been working on and off for some time and always ended up with failure. There are many documentations available on the net however the one that worked for me is using IP trunks and here’s how it is done.

Lets take an example so it’s easy to identify. Let’s assume the the two PBXes are at Sydney & Melbourne.

In Sydney PABX, create a new trunk, name it to-Melbourne. For the peer details set it up as follows:

Trunk Name: melbPABX


In the Melbourne PABX, create a new trunk with a name to-Sydney. For the peer details set it up as follows:

Trunk Name: sydPABX


Points to remember:

  • Password common for both trunks.
  • Trunk names used under Peer Details acts as the usernames.
  • Extension numbers should differ on each PABX otherwise it would not be possible to route calls correctly.

[Asterisk/FreePBX] Blocking incoming calls to an extension

You may have a scenario where you’ve a particular extension assigned to a special service and you want only a select group of people/extensions to be able to make calls to this extension.

This can be easily achieved with the dial plan below. There’s just one file to edit : extensions_custom.conf

exten => _X.,1,Playback(invalid)
exten => _X.,n,Hangup

exten => 7572/7520,1,Answer()
exten => 7572/7507,1,Answer()
exten => 7572/7584,1,Answer()
exten => 7572/4914,1,Answer()
exten => 7572/7500,1,Answer()
exten => 7572/_XXXX,1,Goto(kick,${EXTEN},1)
exten => 7572/_X.,1,Answer()

Let’s analyse what’s happening here.

The first section [kick] tells Asterisk to play a message saying the dialed destination is invalid and then to hang up.

The next section [from-internal-custom] defines what extension can connect/dial to this particular extension (in this example ext 7572 is the one needing incoming restrictions).
The part before the “/” defines the destination extension and the part after the “/” defines the source extensions.
So in this scenario extensions 7500, 7507, 7520, 7584 and 4914 will be able to call 7572. All other 4 digit extensions will be sent to [kick] which will terminate their calls.

The last part will allow any other call to be picked up by the extension (for example, an external call).


Original idea from : source

Asterisk/FreePBX Time Conditions with Multiple Time Zones / Daylight Saving

This solution is for older Asterisk/FreePBX installations (v2.11 and below) where the Time Condition module doesn’t have the Time Zone dropdown.

TZThis is based on a bash script which alters MySQL DB values to represent the change in DST.

This script will help you to automate DST changes easily and and have peace of mind about not having to manually change every year.


PABX is for an Australian company which has offices in multiple states. Main office is in NSW which observes daylight saving whereas a branch office is in QLD which does not observe daylight saving. Server time is set for NSW timezone. Continue reading “Asterisk/FreePBX Time Conditions with Multiple Time Zones / Daylight Saving”

Garmin Vivosmart HR Review – First Impressions and General Use Review

I’ve been hitting the gym more regularly in the last few months and I’ve been looking into getting my first fitness tracker for some time. The main reasons behind the push is that I work at odd hours and was wanting to keep track of my health and general fitness and see where I’m heading to. You could say this this review will encompass many points that someone who is about to take the first steps in getting into the fitness tracking bandwagon.

So without further ado, let’s start with the review.

Garmin Vivosmart HR

What this review is about?

There are plenty of reviews for this device available from review experts, athletes and other users who had been using some sort of fitness tracker. This review will be based on someone who would be thinking of getting a tracker, no prior experience with trackers and who had been working out in the gym for some time (say 6 months to 1 year).

So you will see things that I liked, issues that I faced with this device and in general deciding what tracker to buy and why I went with Garmin’s Vivosmart HR.

I will not be doing scientific measurements/comparisons as there are plenty of other sites that have done it. Just a regular use review.

Continue reading “Garmin Vivosmart HR Review – First Impressions and General Use Review”

Asterisk Phone Inventory / Useragent List

Create batch file with below and run the script. It will list down the Useragent, Contact & Username.

asterisk -rx “sip show peers” | cut -f1 -d/ | grep -P ‘\d\d\d\d’ | grep -vP ‘(UNKNOWN|Unmonitored)’ |
while read PEER
asterisk -rx “sip show peer ${PEER}” |
grep -P “(Useragent|Contact|Username)”
echo “====”

Asterisk/FreePBX – How to restrict an extension to call certain extension only

There may come a time that you want a public access phone that can only dial out a certain set of extensions. To accomplish this, a custom context needs to be created and applied to that extension.

Here’s the scenario.
Extension 312 is the public access phone. This should only be able to dial x530 & x555. All other calls needs to be terminated.

Define the context in extensions_additional.conf

exten => 530,1,goto(from-internal,${EXTEN},1)
exten => 555,1,goto(from-internal,${EXTEN},1)
exten => _.,1,hangup()

If FreePBX, go inside the extension and the context from from-internal->from-intercom. Apply the settings and you’re good to go.