I was able to create simple solution VOIP users call exten 500 which is [meetme] exten => 500,1,Playback,thereare exten => 500,2,MeetmeCount,500 exten => 500,3,Playback,callersin exten => 500,4,Meetme,500|pMs|1234 exten => 500,5,Playback,goodbye exten => 500,6,Hangup
later somebody calls extension 501 which moves 1-test to /var/spool/asterisk/outgoing/ 1-test looks like this Channel: Sip/[EMAIL PROTECTED] (you put whatever you want) Callerid: 1 MaxRetries: 1 RetryTime: 60 WaitTime: 30 Context: common Extension: 500 Priority: 1 exten => 501,1,System(/bin/cp /etc/asterisk/1-test /var/spool/asterisk/outgoing/ ) exten => 501,2,Hangup The problem with this solution is that the 501 needs to be dialed separately. Any ideas how to enable 501 in conference call. Thx > Hello, > > I am thinking about this, > > ----POTS--CONFERENCE-BRIDGE > | > | > | > PSTN > | > | > ASTERISK > | > INTERNET > | > | > VOIP USERS > > Users registers with asterisk, they join the confrence and later (or maybe > at the begining) asterisk automatically (or maybe manually) calls the POTS > conference bridge using the PSTN network. > This would allow all VOIP users to interact with users on the pstn > conference side. Any ideas how this could be done if possible. > > Thanks > > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
