[asterisk-users] invite to conference by a call file

[Dovid Bender]

You would make the call file the same way you are now. 100 was the conf
room ID. Have a look at the documentation how to do it. Also take a look at
the default settings in confbridge.conf

voice1*CLI> core show application ConfBridge

  -= Info about application 'ConfBridge' =-

[Synopsis]
Conference bridge application.

[Description]
Enters the user into a specified conference bridge.  The user can exit the
conference by hangup or DTMF menu option.
This application sets the following channel variable upon completion:
${CONFBRIDGE_RESULT}:
    FAILED:The channel encountered an error and could not enter the
conference.
    HANGUP:The channel exited the conference by hanging up.
    KICKED:The channel was kicked from the conference.
    ENDMARKED:The channel left the conference as a result of the last marked
    user leaving.
    DTMF:The channel pressed a DTMF sequence to exit the conference.
    TIMEOUT:The channel reached its configured timeout.

[Syntax]
ConfBridge(conference[,bridge_profile[,user_profile[,menu]]])

[Arguments]
conference
    Name of the conference bridge.  You are not limited to just numbers.
bridge_profile
    The bridge profile name from confbridge.conf.  When left blank, a
    dynamically built bridge profile created by the CONFBRIDGE dialplan
    function is searched for on the channel and used.  If no dynamic
profile is
    present, the 'default_bridge' profile found in confbridge.conf is used.
    It is important to note that while user profiles may be unique for each
    participant, mixing bridge profiles on a single conference is _NOT_
    recommended and will produce undefined results.
user_profile
    The user profile name from confbridge.conf.  When left blank, a
dynamically
    built user profile created by the CONFBRIDGE dialplan function is
searched
    for on the channel and used.  If no dynamic profile is present, the
    'default_user' profile found in confbridge.conf is used.
menu
    The name of the DTMF menu in confbridge.conf to be applied to this
channel.
     When left blank, a dynamically built menu profile created by the
    CONFBRIDGE dialplan function is searched for on the channel and used.
If no
    dynamic profile is present, the 'default_menu' profile found in
    confbridge.conf is used.

[See Also]
ConfBridge(), CONFBRIDGE, CONFBRIDGE_INFO

On Tue, Mar 20, 2018 at 10:44 AM, Atux Atux <atuxnull at gmail.com> wrote:

> thanks a lot for the reply.
> [call-file-test]
> Exten => 10,1,Answer
>  same => ConfBridge(100)
>
>
> i assume 100 is the conference room, correct?
> where do i write the SIP numbers to invite(internal or external)?
> what about the PIN?
>
>
> On Tue, Mar 20, 2018 at 4:41 PM, Dovid Bender <dovid at telecurve.com> wrote:
>
>> Atux,
>>
>> This should work:
>> [call-file-test]
>> Exten => 10,1,Answer
>>  same => ConfBridge(100)
>>
>> On Tue, Mar 20, 2018 at 10:34 AM, Atux Atux <atuxnull at gmail.com> wrote:
>>
>>> Hi. in my system i have a conference room where someone can call it eg
>>> 698 dial the PIN eg 1234 and enter the room as a user. The admin enters in
>>> through a different number and PIN.  I would like to have a call file and
>>> call all participants eg 610-619 at certain time of the day and give them
>>> access to the conference.
>>> During my try i managed to create a call file where it calls the a SIP
>>> phone and it can hear the monkeys (just for test).
>>> here is the call file
>>> Channel: SIP/601
>>> MaxRetries: 2
>>> RetryTime: 60
>>> WaitTime: 30
>>> Context: call-file-test
>>> Extension: 10
>>>
>>>
>>>
>>> and here is the entry in extensions.conf
>>>
>>> [call-file-test]
>>> exten => 10,1,Answer()
>>> exten => 10,n,Wait(1)
>>> exten => 10,n,Playback(tt-monkeys)
>>> exten => 10,n,Wait(1)
>>> exten => 10,n,Hangup()
>>>
>>>
>>> i did not manage to make it call more SIP phones and invite them to the
>>> conference
>>>
>>> Any ideas please?
>>>
>>> --
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>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
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>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
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