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? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20180320/515f2cf1/attachment.html>
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? > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20180320/5908a2a8/attachment.html>
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? >> >> -- >> _____________________________________________________________________ >> -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20180320/800b487b/attachment.html>
Here I'm using the "Page" application to make a conference call "on the fly". [office] exten => 2222,1,Dial(SIP/desk2,150) ???same => n,Hangup() exten => 3333,1,Dial(SIP/desk3,150) ???same => n,Hangup() exten => 4444,1,Dial(SIP/desk4,150) ???same => n,Hangup() exten => 5555,1,Dial(SIP/desk5,150) ???same => n,Hangup() exten => 6666,1,Dial(SIP/desk6,150) ???same => n,Hangup() ; Conference call exten => 8888,1,Answer exten => 8888,n,Page(Local/2222 at office&Local/3333 at office&Local/4444 at off ice&Local/5555 at office&Local/6666 at office,d) same => n,Hangup() -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20180320/bd1f1af9/attachment.html>
All the aforementioned techniques need change everytime on the dialplan. I need the office secretary to edit a file (call file) and place it in a particular folder in their windows PCs. this folder is the outgoing folder of LINUX shared through samba in LAN. i need to make it as easy as possible, please. On Tue, Mar 20, 2018 at 5:41 PM, Frank Vanoni <mailinglist at linuxista.com> wrote:> Here I'm using the "Page" application to make a conference call "on the > fly". > > > > [office] > > exten => 2222,1,Dial(SIP/desk2,150) > same => n,Hangup() > > exten => 3333,1,Dial(SIP/desk3,150) > same => n,Hangup() > > exten => 4444,1,Dial(SIP/desk4,150) > same => n,Hangup() > > exten => 5555,1,Dial(SIP/desk5,150) > same => n,Hangup() > > exten => 6666,1,Dial(SIP/desk6,150) > same => n,Hangup() > > ; Conference call > exten => 8888,1,Answer > exten => 8888,n,Page(Local/2222 at office&Local/3333 at office&Local/4444 at office > &Local/5555 at office&Local/6666 at office,d) > same => n,Hangup() > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20180322/c13bc6dd/attachment.html>