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??
Dude stop spamming the forum if you want to get a job go to freelancer or PM people.