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 <atuxn...@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 <do...@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 <atuxn...@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? >>> >>> -- >>> _____________________________________________________________________ >>> -- 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 >>> >> >> >> -- >> _____________________________________________________________________ >> -- 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 >> > > > -- > _____________________________________________________________________ > -- 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 >
-- _____________________________________________________________________ -- 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