hello everyone,
please help me with the following scenario.
i am trying to setup an asterisk based solution which will enable a caller to ring in on a predetermined number and connect to literal meeting room which has an audience and a public speaker and be able to share in the audience discussions.
the meeting room has mikes, speaker system and amplifier to manage the sound system. pc (with asterisk stack), adsl connection etc.
my aim is to get sip channel in to sound playback and sound capture to sip channel out. this will enable the caller to hear what is being discussed and when the caller speaks in the meeting room audience will be able to hear the caller.
so my question is. how do i go about achieving this solution?
i tried exten 700,1, dial(console/dsp)
…
but the /var/log/asterisk/message file says cannot re-open or device busy
edited:
(i am aware that app_jack for asterisk 1.6 supports this problem but my distro doesnt support 1.6 for now and i need to get this going soon)
best regards