[Freeswitch-dev] Development Info

[Georgi Stefanov]

 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



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