i am trying to redirect two sip channels to a conference room. it could be possible either through Dialplan (extensions.conf) or using AMI…i know the process flow via dilaplan but in case of AMI how clients (say A and B) communicate via asterisk?
what i know is both clients will open a session with asterisk on port 5038(default), and asterisk administrator will bridge them for voice call…(automatic routing not possible?) i.e both clients will be completely dependent upon Asterisk admin??