custom automated meeting

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Asterisk Users 10 Comments

I need your help in implementing the following scenario:

A certain extension will ring two sip phones simultaneously and when one
of them answers, the other keeps ringing until it answers too, and then
all three (the caller and the other two) are immediately placed in a
conference room (same room for all three).

Can we do it?

10 thoughts on - custom automated meeting

  • I just want to make two specific sip phone sets to ring together, when
    someone dials a specific incoming extension. And then, when each of the
    ringed sets answers, to be placed immediately into meeting session with
    the caller together with the other phone set.

    Here is exactly what I mean:

    Person A dials 123456789. Asterisk routes the incoming call and rings
    sip phones B and C. Person B answers phone B and starts talking with
    person A, while phone C keeps ringing. A minute later, and while A and B
    are still talking together, person C answers phone C, and starts talking
    with A and B together (that is aromatically all being placed in the same
    conference session).

    Is that doable?

  • One way to do this (there are probably more and better ways). Incoming call
    to 123456789 launches meetme(1234,b(connecta.agi))
    Connecta.agi calls lines B and C and connects them to meetme(1234).

  • on 11/01/2011 03:25 PM Danny Nicholas wrote the following:

    Thanks, but could you be more elaborate please?
    Where can I find connecta.agi ?

  • There is only one way to find out connecta.agi in /var/lib/asterisk/agi-bin
    directory by developing it yourself.

  • Although if you dig through the archives you can find a good cross-section
    of AGI samples. Check the Asterisk Cookbook wikis as well.

    [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Sammy Govind
    Sent: Tuesday, November 01, 2011 9:08 AM

    There is only one way to find out connecta.agi in /var/lib/asterisk/agi-bin
    directory by developing it yourself.

    on 11/01/2011 03:25 PM Danny Nicholas wrote the following:

    call

    Thanks, but could you be more elaborate please?
    Where can I find connecta.agi ?

    the
    B

  • You need simple dialplan of four steps:
    same =>n,Set(conf_name=conf-${RAND(1,1000)})
    same =>n,Originate(SIP/dev1,app,MeetMe,${conf_name},dFI1x)
    same =>n,Originate(SIP/dev2,app,MeetMe,${conf_name},dFI1x)
    same =>n,MeetMe(${conf_name},dFI1xAC)
    same =>n,Noop(do post conference stuff)

    2011/10/31 Thanasis :

  • on 11/01/2011 05:41 PM Yaroslav Panych wrote the following:

    Thanks!
    What is the meaning of the options dFI1xAC passed to
    app,MeetMe,${conf_name} ?
    Where can I find them described please?

  • on 10/31/2011 11:59 PM Thanasis wrote the following:

    FWIW, using call files:

    Here is the relevant section of the dialplan:

    exten => 300,1,Noop(creating conference)
    same => n,Set(conf_name=conf-${RAND(1,1000)})
    same => n,System(/etc/asterisk/scripts/callgenerator SIP/dev1 ${conf_name})
    same => n,System(/etc/asterisk/scripts/callgenerator SIP/dev2 ${conf_name})
    same => n,MeetMe(${conf_name},dFI1xAC)
    same => n,Noop(do post conference stuff)

    … and here is the script /etc/asterisk/scripts/callgenerator:

    #!/bin/bash
    PHONE=$(echo $1 |cut -f2 -d”/”)
    ROOM=$2
    echo “Channel: $1” > /var/spool/asterisk/tmp/${PHONE}.call
    echo “Application: MeetMe” >> /var/spool/asterisk/tmp/${PHONE}.call
    echo “Data: ${ROOM},dFI1x” >> /var/spool/asterisk/tmp/${PHONE}.call
    mv -f /var/spool/asterisk/tmp/${PHONE}.call /var/spool/asterisk/outgoing

    PS: Thanks much to Yaroslav for his help 🙂