One Touch Recording??

Lost

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Been pulling my hair out trying to get record on demand to work. It works if I dial 7777 and press *1. It shows up as "from-did-direct" in call monitor and I can record my own MOH. However when making calls *1 doesn't work, (from-internal in call monitor).

General settings are: trwW and TwW extensions are set to "record on demand" using xlite softphones. (Record always on DOES work!).

I've added the courtisey beep and I can hear it when I press *1 and set the digitfeaturetimeout=1500

My sip_custom.conf looks like:

[general]
context=unauthenticated
allowguest=no
srvlookup=yes
udpbindaddr=0.0.0.0
tcpenable=no

soft-phones](!)
type=friend
context=xlite
host=dynamic
nat=yes
secret=secret
dtmfmode=auto
disallow=all
allow=h263
allow=gsm
allow=ulaw
allow=alaw

[10](soft-phones)
[11](soft-phones)

My extensions_custom.conf looks like:

[xlite]
exten => _x.,10,1,Dial(SIP/xlite)
exten => _x.,11,1,Dial(SIP/xlite)

Running PIAF 1.7.5.6 purple
Asterisk 1.8
FRPBX 2.9

Any help would be appreciated.

Newbie.:rolleyes:
Auckland
New Zealand
 

Lost

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I've already checked /var/spool/asterisk/monitor and what shows up there simply reflects what shows up in the in the GUI under call monitor.
 

RJMiller

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Did you ever resolve this? I also have this problem. I have tried to ignore it but the users want it (even though I see it becoming a legal problem when they stick a customer's nose in it). FOP2 does the recording properly and even puts it in the voicemail if you set it up for this. Still seems that if it is supposed to work it should.
 

rossiv

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It works for me - the recording, at least. Web GUI is a FreePBX issue. GUI recordings work for a few extensions, but not for others. Never bothered to figure it out.
No one has posted any call logs or status outputs or anything. Please post those and all the info requested in the thread in my signature.

What mad On Demand (*1) Call Recording work for me was (found it) doing this:
1. Add "featuredigittimeout=3000" and "courtesytone-beep" to /etc/asterisk/features_general_custom.conf
2. Set Asterisk Dial command options in FreePBX General Settings to "tTrwW" and Asterisk Outbound Dial command options to "tTwW"

Speaking of, what are your dial options?
 

rossiv

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So I think I have found an Asterisk naming convention error or FreePBX bug or something.
For example, I called 8432222222 from extension 215 and pressed *1 to record the call. The beep played and it showed up on Asterisk CLI. When the call was ended, the file was placed in /var/spool/asterisk/monitor. BUT it didn't show up in FreePBX. I looked back and there was only one file with 8432222222 in the name but it did not say 215 where the extension usually us, but my outbound CID number, 843235XXXX. So I renamed the file from
Code:
auto-1316954994-843XXXXXXX-8432222222.wav
to
Code:
auto-1316954994-215-8432222222.wav
and BOOM! I get an icon in the Call Monitor panel of FreePBX.
RecordingCallMonitor215.jpg
 

wardmundy

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Definitely a bug. Please open a FreePBX ticket. Thanks.
 

paulnye

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Need this feature too

My users are asking for this feature too. Cant wait for a bug fix.
 

EndeavorPBX

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I've been told that they are completely re-writing the code to support this feature in 2.10 and moving away from the Asterisk-based code that supports this now. So, keep your fingers crossed, and download the 2.10 beta so that the dev team can get your input!
 

paulnye

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I've been told that they are completely re-writing the code to support this feature in 2.10 and moving away from the Asterisk-based code that supports this now. So, keep your fingers crossed, and download the 2.10 beta so that the dev team can get your input!

What do I download, and where? My freepbx is 2.9
 

rossiv

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What do I download, and where? My freepbx is 2.9
Slllooowwww down there. FreePBX 2.10 is still WAY in Beta from what I have read and seen right now. It will come in its time.
Edit: And even if you did download it now, many (if not most) PIAF modules would probably not work.
 

ostridge

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What made On Demand (*1) Call Recording work for me was (found it) doing this:
1. Add "featuredigittimeout=3000" and "courtesytone-beep" to /etc/asterisk/features_general_custom.conf
2. Set Asterisk Dial command options in FreePBX General Settings to "tTrwW" and Asterisk Outbound Dial command options to "tTwW"

Sorry to reply to an old post but just to update:
Code:
;! this file is  features_general_custom.conf
featuredigittimeout=3000 ; Max time (ms) between digits for
                        ; feature activation  (default is 1000 ms)
    ;( ##by the way this set to =5000 ms in followme.conf ##)
 
;courtesytone=beep     ;not needed this is already in features_general_additional.conf

Also for Asterisk11 -hmm I think it now is the xX dial options that apply with automixmonitor see https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial
"w - Allow the called party to enable recording of the call by sending the DTMF sequence defined for one-touch recording in features.conf.
W - Allow the calling party to enable recording of the call by sending the DTMF sequence defined for one-touch recording in features.conf.
x - Allow the called party to enable recording of the call by sending the DTMF sequence defined for one-touch automixmonitor in features.conf.
X - Allow the calling party to enable recording of the call by sending the DTMF sequence defined for one-touch automixmonitor in features.conf."
Regards
Also For Asterisk 11 (from digium )
w - Allow the called party to enable recording of the call by sending the DTMF sequence defined for one-touch recording in features.conf.
W - Allow the calling party to enable recording of the call by sending the DTMF sequence defined for one-touch recording in features.conf.
x - Allow the called party to enable recording of the call by sending the DTMF sequence defined for one-touch automixmonitor in features.conf.
X - Allow the calling party to enable recording of the call by sending the DTMF sequence defined for one-touch automixmonitor in features.conf.
bye
 

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