Thanks for your efforts.
(Mybe my problem is lack of knowledge rather than a real problem. But anyway please consider and help)
Does anybody used this macro? I am trying to use it on a * 1.0.10 and it does not seems to work. when I see the flow of commands in a sip debug mode, the 'Cut' command does not do anything.
And after all is there a good solution for conferencing on Asterisk? I have only encountered some statements about this title here and it seems there is no real attention for call conferencing in asterisk.