Call recording / automon

Hi All,

I’ve inherited an asterisk 1.4.16.2 machine. I’m completely new to this and I’m struggling with the automon feature. After googling a bit I found out that I should enable automon in features.conf and add the wW flags to the dial command. However when I press *1 during a call nothing happens in the asterisk -r screen and no file is created in the /var/spool/asterisk/monitor directory.

My features.conf contains the following section and DYNAMIC_FEATURES is commented out:

[featuremap]
;blindxfer => #1 ; Blind transfer (default is #)
;disconnect => *0 ; Disconnect (default is *)
automon => *1 ; One Touch Record a.k.a. Touch Monitor
;atxfer => *2 ; Attended transfer
;parkcall => #72 ; Park call (one step parking)

This is the dialing macro , where I have added the wW flags (the r was already there):

[macro-pandial]

#This is a general dial macro with Call Waiting function.
#Used for all regular SIP users:

exten => s,1,GotoIf($["${ARG1}" == “SIP/Reception&SIP/Wireless”]?OK:2)
exten => s,2,Set(STATUS=${DEVSTATE(${ARG1})})
exten => s,3,GotoIf($["${STATUS}" == “NOT_INUSE”]?OK:WAITING_CALL)
exten => s,n(WAITING_CALL),Set(VMEXTENSION=${MACRO_EXTEN})
exten => s,n(WAITING_CALL),Background(callwaiting)
exten => s,n(WAITING_CALL),WaitExten(5|m)
exten => s,n(WAITING_CALL),Dial(${ARG1},10,rwW)
exten => s,n(WAITING_CALL),Goto(s,2)

exten => s,n(OK),Dial(${ARG1},20)
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})
exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN})
exten => _s-.,1,Goto(s-NOANSWER,1)

#If user press 0 or * she will be transferred to main IVR.
exten => o,1,Goto(from-pstn,1234567,1)
exten => a,1,Goto(from-pstn,1234567,1)

What am I missing here? Please advice!

Thanks!

One addition: it seems that the automon feature is is working correctly since a ‘feature show’ outputs:

Builtin Feature Default Current


Pickup *8 *8
Blind Transfer # #
Attended Transfer
One Touch Monitor *1
Disconnect Call * *
Park Call
[…]

Anyone? :smile:

I tried enabling this feature just now to help troubleshoot and i can not get mine to work either.

I noticed you are not using the command SET(DYNAMIC_FEATURES=automon) before your dial command, but perhaps that is because you have it defined globally.

Now I am interested in a solution to this issue as well.

In my setup Asterisk does not seem to notice me dialing *1 while i am on a call. I know dtmf is working ok because i have a voice menu system that is working fine…

Here is the important part of my extensions.conf:

exten => _66XX,1,answer()
exten => _66XX,2,Set(DYNAMIC_FEATURES=automon)
exten => _66XX,3,Dial(SIP/${EXTEN},5,wW)
exten => _66XX,4,Voicemail(${EXTEN}@default)