Call recording

Dear all

Problem: Unable to record call after routing a call to the sip agent.

I used manager api commands to originate, transfer and record the call.

The file gets created but the conversation does not get stored on the file which was created.

The command which i used to record the conversation is the following

Action: Monitor
Channel: sip/brekekeout-080808009(Respondent line)
Format: gsm
Mix: 1

Please help me to sort out the problem

Regards
Karthik

Hi

I was successful in recordign the conversation when i try to first connect to agent then to respondent. In this case also i issued the monitor command once the channels gets bridged.

please help

Regards
Karthik

Hi
I’m trying to record calls, but I don’t get it. What I’d like to do is that the caller or the called could start the recording by pressing *1. I read something about it but I don’t know how extension.conf should look like. Should I modify any other file?

Thank you all.

Have you looked at the features.conf ? and voip-info.org voip-info.org/wiki/index.php … tures.conf

Ian

Thank you very much for your answer. I read the information you referred me to. I saw the option ‘automon=>*1’ in features.conf. Only by uncomment it I will be able to record a call just by pressing *1? Is that all I must do?

Thanks.

Hi,

to record calls I do in extensions.conf:

[IncomingCall]
exten => s,1,Set(CALLFILENAME=${EXTEN:1}${CALLERIDNUM})
exten => s,2,Monitor(wav,${CALLFILENAME},m)

and in /var/spool/asterisk/monitor/ you find all your CALLFILENAME Without do anything in features.conf.

In my configuration i record all the call (Incoming and outgoingCall) and if you have any questions I can help you.

And please if you read my Messsage (Subject “send sms with asterisk”) I’m awaiting your kind reply. :blush: :frowning:

                                                            Sanaâ

Hi Sanaâ,
Thank your for your answer, I’ll try what you suggested. Unfortunately I’m too new in the asterisk world and I need to study a lot more. If I find an answer to your problem I’ll tell you.

See you!

Hi All,

   How to record a conversation after the conversation has started?
   Can we achive it with Manager API instead of using features.conf?

Regards
Karthik

When I record my calls all I get is empty wav files… at some point I modified the extensions.conf file and got to record the ringing but not the call itself, can someone please point me in the right direction? my script works fine on an * 1.2.3 box, but not on * 1.4.0

Hi rantsh,

      Can you say what modifications you have made for recording?

I have got the similar problem. The file gets created but the content doesnot gets stored.

Regards
Karthik

[quote=“dinesh009”]Hi rantsh,

      Can you say what modifications you have made for recording?

I have got the similar problem. The file gets created but the content doesnot gets stored.

Regards
Karthik[/quote]

Hi dinesh009, I’ve written all my advances and problems in this thread in the quest to get this thing working properly, hopes it can help you, and if you solve the issue please let me know how

This is the http://forums.digium.com/viewtopic.php?t=16283

Hi, I was finally able to record calls. In extensions.conf I have:

exten=>2030,1,Set(MONITOR_EXEC=/etc/asterisk/soxmixwav.sh)
exten=>2030,2,Set(CALLFILENAME=/var/spool/asterisk/monitor/${EXTEN})
exten=>2030,3,Answer
exten=>2030,4,Monitor(wav,${CALLFILENAME},M)
exten=>2030,5,Dial(SIP/2030,10,Ttr)
exten=>2030,6,Hangup

With this I get two different files in /var/spool/asterisk monitor
Typing soxmix file1.wav file2.wav out.wav I get a file with whole conversation.

I’d like to start recording the call when either the caller extension or the called extension dial *1. I’ve tried what I’ve read but I don’t get it to work. Any suggestion will be apreciated.

Thank you.

1 Like

Is there a way to get the call to record automatically without either party pressing W or w?

I am following same but my files are only 44 KB and nothing records …any help ?

This thread have 11 years, a lot thing has been changed please open a new one