[asterisk-dev] channel driver question

[Gallmeier, Jonathan]

Thanks for your reply and sorry that I posed a very general question. I've got a ton of verbose messages in my code and nothing seems to point to the problem. I didn't think about the sip side. I'll look at the sip debug messages and see if that points me in the right direction. If I saw where the hangup originated, I could figure out the problem and fix it. :)

I think my real problem is my lack of understanding how bridging works, and how the channel driver works to some extent. I'm not sure what flavor of bridging I'm using. I've left the bridging to the default asterisk routines. Basically I've set my dial plan up to accept a call on a given number from the sip phone. It then turns around and dials an address via my channel driver using the Dial command. I understand the apps... still spinning up on the channel driver side.

Thanks again for your reply... 

Jonathan


> -----Original Message-----
> From: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-
> bounces at lists.digium.com] On Behalf Of Martin
> Sent: Thursday, September 10, 2009 7:11 PM
> To: Asterisk Developers Mailing List
> Subject: Re: [asterisk-dev] channel driver question
> 
> 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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