Thanks Steven, I am looking the DOCs too, Currently I succeeded to place my non-sip call into the dialplan. I see the dialplan execution is started but when playback execution is started is stops with hanging up my call. ( I have tried to pre-answer but...) I need something like " your_io_routines.outgoing_channel is called create a new session (with switch_core_session_request()) parse outbound_profile->destination_number to get called number qualify the session, attached channel and attached private data The channel is currently in CS_NEW Change channel state to CS_INIT Make the phone ring (and call switch_channel_mark_ring_ready()) your_state_handler.on_init is called Change channel state to CS_ROUTING your_state_handler.on_routing is called What to do here? The call is answered switch_channel_mark_answered(channel) " the info from Authoring_Freeswitch_Modules but with more details In my case: I have no calling of your_io_routines.outgoing_channel I do the codec staff in on_init function On_Routing I do the pre-answer On_Receiving I try to start rtp session I can not understand am I doing it right or not ? (probably not, but I need to do it the right way, but I do not know the right way) So every kind of info is welcome ----------------------------------------------------------------- Намери автомобил от най-бързоразвиващия се сайт за коли в България! http://www.carmarket.bg/