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FOP2 Community Forum

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silas2

@silas2
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Recent Best Controversial

  • Duplicate Logins
    S silas2
    Support

    If you have a real-life scenario where you want the manager of the switchboard operator to login as the switchboard operator to oversee, i.e. multiple logins with the same extension, can you do this?
    When I try this, one of the swtichboards seems to get 'locked', i.e. doesn't refresh correctly. I don't know if this is my configuration error or just by fop2 design?


  • Can't get list of Conference participants
    S silas2
    Support

    Now I get it. I can see the list now. I didn't realise you have to get the conferencing working thru asterisk first, I thought fop2 did the conferencing.


  • Can't get list of Conference participants
    S silas2
    Support

    ...and set up all the MeetMe exten's in the dailplans?


  • Can't get list of Conference participants
    S silas2
    Support

    Have I got to configure the meetme.conf in asterisk?


  • Can't get list of Conference participants
    S silas2
    Support

    I don’t seem to get the list of participants depicted in the online help or understand how to invite anyone to a conference.
    Here is my fop2.cfg:
    ;Users
    user=103:secret1:meetme
    user=104:secret2:meetme
    user=503:secet3:all:MotherHen
    ;Groups
    group=MotherHen:SIP/503,SIP/103,SIP/104,CONFERENCE/1000
    (The contexts for all the users and the conference are allowing normal internal extension-extension calls)
    Is this right?


  • Automatic recording
    S silas2
    Support

    I want to record all calls which go thru a certain dialplan (failing that, all calls)
    Is there any way to configure fop2 to do this, or do I have to go thru asterisk?


  • Can't Listen In
    S silas2
    Support

    Thanks Nicolas, it was my fault, there was no dialplan set up incoming calls for the listening phone, whoops! It is getting an incoming/listening call now. (I'm an Asterisk newbie).


  • Can't Listen In
    S silas2
    Support

    It is (buttons.cfg) generated by code, but the code was written by hand so it may still be wrong!
    They are a minimum, e.g.

    [SIP/107]
    type=extension
    extension=107
    context=TeamMember
    label=Pauline Ford

    And the calls are going thru ok, and they light up fop2 buttons ok, so you would think the buttons.cfg is matching the sip.conf.
    Do I really not need a dialplan/extenstion.conf entry for the listen in?


  • Can't Listen In
    S silas2
    Support

    I haven't set any privacy options for the buttons, I guess the default is off?
    I have set the manager.conf to
    read= all
    write=all
    The phone listening in is not receiving any incoming call event. Could it be the extensions.conf/dialplan, I haven't done anything to enable that for listening in?


  • Can't Listen In
    S silas2
    Support

    I can hangup the call, so I guess the permissions are getting thru ok, maybe(?)


  • Can't Listen In
    S silas2
    Support

    I'm just trying to listen in to ongoing call.
    Ok, I realise I was misreading the help, I've switched back to spy_options="bq", I'm still unable to listen in.
    Any other suggestions?


  • Can't Listen In
    S silas2
    Support

    Asterisk 1.8, fop2.
    I’ve put spyoptions=4 in appropriate button in buttons.cfg

    • permissions set to ‘all’ for the user entry in fop2.cfg ( I don’t know if this is granting the same permission twice)
    • I’ve put force_parameter_delimiter="," spy_options="4" in fop2.cfg
    • read = system,call,command,agent,user,originate
      write = system,call,command,agent,user,originate in Asterisk manager.conf,
      but sill no listen in!!!
      What else to try?

  • Remotly Trigger a Reload
    S silas2
    Support

    What is the best way to trigger a reload remotely (say from a Windows Service in my case)?


  • SIP.conf/fop2 users, how do you tie them up?
    S silas2
    Support

    Thanks, that was it.


  • SIP.conf/fop2 users, how do you tie them up?
    S silas2
    Support

    Is it that the buttons bind to channels, which take the form SIP/username (=sip.conf username??)
    (link to relevant documentation would be welcome if I'm overlooking the obvious)


  • SIP.conf/fop2 users, how do you tie them up?
    S silas2
    Support

    If know I must be exposing a fundamental misunderstanding, but if asterisk has all these sip calls getting 'bridged' between the registered sip phones and the outside world, doesn't fop2 somehow 'sit' on top of that to flash the lines which are active, etc, etc?


  • SIP.conf/fop2 users, how do you tie them up?
    S silas2
    Support

    I'm sure this is staring me in the face, but I don't quite understand the relationship between the user list in fop2.cfg and Sip.conf user list/sip registrations , how do you tie them together?(they are presumably the same thing(?))


  • Linux + fop2 newbie, install issue
    S silas2
    Support

    Thanks, that fixed it.


  • Linux + fop2 newbie, install issue
    S silas2
    Support

    Thanks for getting back to me.
    I tried the uname -a, which returns i686, which I think means 32 bit, and I downloaded the file: fop2-2.20-debian-i386.tgz which I guess is the correct one, and ran the chmod a+x, but still the 'Command Not Found', any further suggestions?


  • Linux + fop2 newbie, install issue
    S silas2
    Support

    As far as I can tell I've completed all the installation steps on Ubuntu, I go to run the fop2_server --test and I just keep getting 'command not found'. I do a Alt-f2 to run the app, even though I'm not sure its running as I can see it listed if I run ps -A. Where am I going wrong?

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