AlarmReceiver with Asterisk 13


I have read numerous stories of people discouraging others to use AlarmReceiver with recent versions of Asterisk.

It took me days to make it work, but it now works reliably.

Here is my setup :

  • Asterisk 13.0.0 on a Raspberry Pi with Archlinux
  • Alarm connected to Line 1 of a Linksys PAP2T adapter (hw 0.3.5 fw 5.1.6)

Extensions.conf :
exten => 123,1,NoOp(Alarm received)
exten => 123,n,Answer
exten => 123,n,AlarmReceiver()
exten => 123,n,Hangup()

Panel is a DSC PC1832 v4.6 NA, programmed to dial phone number 123 with Contact ID protocol.

DTMF is inband, both in the PAP2T configuration and in sip.conf of Asterisk.

In asterisk.conf I have mindtmfduration = 20 (not sure if it’s required but the default is written to be 80 and the contact-id DTMF pulses are 50ms long…)

Most importantly here is my alarmreceiver.conf :
timestampformat = %a %b %d, %Y @ %H:%M:%S %Z
eventcmd = /home/pi/
eventspooldir = /alarm
logindividualevents = no
fdtimeout = 4000
sdtimeout = 4000
loudness = 600

(change the directories accordingly to your setup)

[color=#FF0000]The CRITICAL setting here is “loudness”[/color]. Too high and the handshake is not detected by the panel, and nothing will work !!! Took me some trial and error to find that one out and I suspect it’s because of this that people report that AlarmReceiver doesn’t work…

I connected another phone to the line to listen in when the panel was calling. With a value too high for “loudness”, you can hear the tone, but the panel doesn’t react by sending the DTMF pulses. The natural reaction is to increase the volume… but no ! With a lower volume, it works !

Just wanted to share this information somewhere so that people who will Google it in the future will find it (hopefully). Have fun.



PS: By the way, because it seems like signal levels are important, make sure to check the impedance settings in the VoIP adapter. If you use a PAP2T, you have a dropdown in the advanced config.
I have set 600 ohms (impedance in north america); I have set the FXS output and input gain to -3 and the FXS power level limit to 8.

Hope this helps

Excellent post :fu: … just fixed my Alarmreceiver settings by changing the loudness from default 8192 to 768. This works - slightly above the lowest value that operates OK - perfectly for me.

I measured signal result with a RigolScope agains a Visionic alarmpanel. Strangely enough one should NOT clearly hear/see the loudness of the KissOff tones. With the loadness value between 700-1200 my AlarmCalls are handled without any retry, even when I fiddle the telephone line.

Loudness value 100 up to 300, timed-out and/or was not understood at all. Any value above 1536 was followed by multiple signal retries during single call which causes a longer duration of alarm processing.
See this, though OK this single call was retried , indicating by 3 small/lowvalue KissOff invitation retries:

Note: The call starts with a 5 digit phonenumer followed by 3 retry invitations and 2 retries for the alarmcode. Total duration is about 20 seconds.

The default (8192 and also others starting from 4096) often resulted in missed alarms and multiple calls to retry the status and randomly finished to a communication failure.

Now my call is handled in about 7 seconds, see below for picture: fails as I’m a new user… I can only upload a single picture… Rest assured, after setting the loudness to 768 the call did not required/produced retries.



Just for completeness, and before I remove the picture… below my successful signal quality after I changed the AlarmReceiver value from 8192 to 768. Up to now all my alarm-status are handled in 7 seconds in a single call with a single code.

Note the call start by dialing (for me a 5 digit phonenumber) followed by a silent handshake and a 16-digit alarmrespons. In the picture this is shown as 5 peaks, some “silience” and 16 other peaks. The bllue line displays call-active.

Hope this will help others.

I have no use for Alarmreciever but that’s a way cool looking graph, What’s it from?

@ johnkiniston : Thx, its a screen-copy of a Rigol Oscilloscope to monitor/measure the (analog) phone-line of the Alarmsystem connected via an ATA to the Asterisk PBX network.

When I would enlarge the signal, one could see the actual DTMF dial and code patterns. Very handy when one need to analyze signal quality.

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Very cool, I was thinking it was a IP Telephony tool I wasn’t familiar with showing the RTP of the call.