[asterisk-users] Inline record

Adrian Marsh Adrian.Marsh at ubiquisys.com
Wed Jun 20 06:22:54 CDT 2007


Ah...

One question though -  Obviously doesn't work for Meetme..  I know I can pre-program meetme to record conferences, but I don't see how to let users start the record on-the-fly.

Nothing at http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe  seems to suggest it can be done..

Can it?

A.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Adrian Marsh
Sent: 20 June 2007 10:38
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Inline record

Scrap that... Tried the Set() method and it worked, so then I moved it from
[general] to [globals] and it does now record the calls.


A.

-----Original Message-----
From: Adrian Marsh 
Sent: 20 June 2007 10:06
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Inline record

Hi Rob, (and Drew)

Thanks for that info, it helped a lot.
I've edited featuremap as detailed, and "show features" gives:

ubiphone*CLI> show features
Builtin Feature           Default Current
---------------           ------- -------
Pickup                    *8      *8
Blind Transfer            #       #
Attended Transfer
One Touch Monitor                 *1
Disconnect Call           *       *


I've added the variable to [general] (although I think it should be "="
instead of "=>" according to the docs, and I've modified my Dial string to:

exten => _6.,3,Dial(${TRUNK2}/${EXTEN:1},,wW) 


But on an call, I still although the DTMF is heard, it doesn't do anything
that I can tell: (numbers hidden)


Everyone is busy/congested at this time (1:0/0/1)
    -- Executing Dial("SIP/227-08865c90", "IAX2/ubigradout/***********||wW")
in new stack
    -- Called ubigradout/************
    -- Call accepted by 193.111.201.75 (format ulaw)
    -- Format for call is ulaw
    -- IAX2/ubigradout-16385 is ringing
    -- IAX2/ubigradout-16385 is making progress passing it to
SIP/227-08865c90
    -- IAX2/ubigradout-16385 stopped sounds
    -- IAX2/ubigradout-16385 answered SIP/227-08865c90
Jun 20 09:45:40 DTMF[22469]: channel.c:2350 ast_write: IAX2/ubigradout-16385
: *
Jun 20 09:45:40 DTMF[22469]: channel.c:2350 ast_write: IAX2/ubigradout-16385
: 1
    -- Hungup 'IAX2/ubigradout-16385'

I'm expecting to see something about recording, and then a file to appear in
the "monitor" or "recordings" directory.

I've restarted A*k as well..  I'll try playing with which keys to use and
see if it's a dtmf issue..

A.


________________________________________
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Rob Schall
Sent: 19 June 2007 19:48
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Inline record

In the features.conf file, under featuremap, add automon => *1

Then in extensions.conf...
[general]
DYNAMIC_FEATURES=>automon   ; Auto Monitor Calls by pressing *1

now if you press *1 while on a call, it will begin recording. Press *1 again
and it will complete the recording.

Rob


Drew Gibson wrote: 
Adrian Marsh wrote:
  
Hi All,

Is there a way to have A*k record calls on-the-fly, at the users
request?  i.e. a possible scenario:

Party A calls Party B
During the call, Party A wants to start recording the call, so presses
"*", A*k announces "recording.." and starting MixMonitor to a file.
Once the call is finished, then A*k emails a copy of the .wav file
over...

I know that meetme can record calls, and I've been able to record calls
from the beginning using Record and MixRecord,  but can't see with Dial
how you'd have A*k listen for the *.

I know that voicemail can email saved messages

So I'm guessing this is a mix of the two..

Cheers,

Adrian

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automon
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+features.conf


  




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