but i 've get new problem,
in my asterisk cli,, he said,
--Got SIP response 482 "Loop Detected" back from 192.168.3.2
>Channel SIP/101-09c402e8 was never answer
any idea,, how to solve it??
i have one question,, client use zimbra from his browser (mozilla/http) and when he call another client his use browser too.. and my question,, what voip can works over http??? because i see in my asterisk 1.6 cli, he use RTP (real time protocol)..??
what i need addons???
i hope someone can solved my problem... pleaseeeeeeeeeeeeeee....... :)