Call recording from cell phone

Discussion in 'General' started by kmullen, Jan 4, 2010.

  1. kmullen

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    I have several users (extensions) that only use the system to forward their extensions to their cell phones. However, recently they would like to have the ability to record some of their phone calls.

    Is there a way for the user to call into the PBX, and make an outbound call "FROM THEIR EXTENSION"?

    This is a little different from DISA, as each user would want to have access to their own recordings on their call monitor GUI.
     
  2. dicko

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    From the generl settings page (and it's hover-over help)

    Asterisk Dial command options: t: Allow the called user to transfer the call by hitting #
    T: Allow the calling user to transfer the call by hitting #
    r: Generate a ringing tone for the calling party
    w: Allow the called user to start recording after pressing *1 (Asterisk v1.2)
    W: Allow the calling user to start recording after pressing *1 (Asterisk v1.2)

    so adding that Ww might be easier.
     
  3. kmullen

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    I appreciate the answer but I am not clear how this would be used.

    I have those options selected already in my dial commands.

    I call into main number and access DISA and get a dial tone. Then what?

    Can you give me a step by step procedure?
     
  4. dicko

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    No just have it the way it is and have the users who want to record a call just dial *1 on their cell-phone.
     
  5. kmullen

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    That works if the user has received the call via follow me without an issue.

    But what if the user is calling into the PBX, then making the business call from the pbx? How does the PBX know where to put the recording?

    I'm missing something here or I haven't explained it very well.

    User Cell Phone <> PBX <> DISA <> Outbound call.

    or

    User Cell Phone <> PBX <> Ext <> Outbound call (how does the user log into extension to make call?
     
  6. dicko

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    I imagine it would but the call in /var/spool/asterisk/monitor/ which is the default location, if you look in that directory, you will find ite easy to decode the nomenclature used, it would not be able to guess the Users extension without help from you though.

    you can write a script to modify the name of the file based on callerid and a lookup table or perhaps a variable that gleans the user's extension using getDTMF() and insert it into "general Settings/Run after record" : here is the hover-over-help

    Run after record: An optional script to be run after the call is hungup. You can include channel and MixMon variables like ${CALLFILENAME}, ${MIXMON_FORMAT} and ${MIXMON_DIR}. To ensure that you variables are properly escaped, use the following notation: ^{MY_VAR}

    or conversely you can have an inbound route based on CID that goes to a custom-context that can do almost anything you want it to.

    Unfortunately you will have to write either of those yourself as it is not part of FreePBX, but the open nature of FreePBX easily enables such extensions of function.
     
  7. kmullen

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    Ok.

    I have tried this several different ways and the calls aren't being recorded. I have listed the methods below.

    1) Call in to main number and try to press *1 before going to DISA. I get "That is not a valid extension"

    2) Call in to main number access DISA, press *1 at dial tone before dialing outbound number. That doesn't work.

    3) Call in to main number access DISA, dial outbound number, press *1 while outbound call is ringing.

    4) Call in to main number access DISA, dial outbound number, press *1 after callee has answered.

    I completed all of these attempts while watching the asterisk CLI, at no time did I see my key presses hitting the server. Call recording works by demand on all extensions, it must be something to do with being within the DISA.

    That's why I was wondering if there is a way to access and dial out from my extension over the cell phone. That would work.
     
  8. dicko

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    *1 will only work after a call is in progress and the audio is passing between the two endpoints.

    To debug add dtmf to the full=> line in /etc/asterisk/logger.conf, restart the logger rasterisk -x "logger reload". any dtmf events will then be sent to /var/log/asterisk/full
    If they don't show it's possible that you might need to mess with dtmfmode= (maybe inband) on your SIP trunks depending on how your VSP operates.

    If all else fails, then a custom-context based on info in:

    http://www.voip-info.org/wiki/view/Aste ... tures.conf

    might allow you to push the name of the monitor file such that it will "appear" in the various GUI's extension reports.
     
  9. johnme

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    Hello

    I need that to.
    Last time i was trying to make it work i realise that elastix do all the job but when you hang up the call it says that 'cant find user in database'.
    So elastix do the recording but at the end it doesnt save anything because the user is unnown.

    If some way elastix new the user then that's it.
    How can we put our mobile phone number in the list of asterisk users?

    All the disa calls are from my cell phone to gsm mv370 then disa to isdn.

    Mv370 is a SIP Trunk and in peer and user details i have put record in and out always.
     
  10. beverlyyan

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    There should be a possibility for the phone call forwarding just like that. You just need some settings for your phone.
     
  11. johnme

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    Hello

    You said 'You just need some settings for your phone'
    So give an exable or something. What do you mean.

    John
     

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