[asterisk-dev] channel driver question

[Martin]

Well it's not that difficult to debug what is going on ... actually if
you're developing the channel driver.
Just poke around and go through the process.... turn on more messages
... add ast_verbose's if you need to and find it ...

You didn't say if you are trying to do native bridging or just
bridging ... It's also possible it's the destination SIP device that
hangs up on you ... did you check sip debug to make sure it's Asterisk
disconnecting the two legs ?

Your question is too general anyways ...

Martin

On Thu, Sep 10, 2009 at 1:18 PM, Gallmeier, Jonathan
<Jonathan.Gallmeier at polycom.com> wrote:
> Hi,
>
>
>
> I’m working on a channel driver and have based my code off of the
> chan_usbradio.c and chan_sip.c. My intention is to bridge SIP to my channel
> driver.
>
>
>
> It _mostly_ works. I can bridge an audio SIP call to my channel and get a
> connection up _until_ I start passing audio RTP data via the read function.
> When I start to pass audio data, Asterisk hangs  up the call. I’m not
> sending null frames in my read function and I’m at a loss as to why the
> channel gets hung up. If I don’t send any data by not registering the file
> descriptor, the call stays up fine, but without the desired audio.
>
>
>
> Any suggestions on how I can debug this???? Do I need to send some special
> frame to make this work correctly?
>
>
>
> Jonathan
>
>
>
>
>
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