[asterisk-dev] Questions regarding a new channel driver

[Earle Clubb]

Hello all,

I'm in the process of writing a channel driver and there are a couple things
I don't quite understand.

1) What is the purpose of bridging?  I have seen that the SIP driver will
perform a native bridge when making a call between two SIP devices.  Is
bridging required in my driver if I'll never make a call between two like
devices?

2) I have call setup and teardown working for my devices, and I have audio
flowing from Asterisk to my devices.  I'm having difficulty delivering audio
from my devices to Asterisk.  I am not using RTP.  I'd like to use the read
callback specified in the ast_channel_tech structure, but I haven't been
able to figure out how to get Asterisk to call the read callback.  I've
looked at other drivers, but I'm just not seeing it.  Right now I've
implemented an io context to listen to my audio port in the same way as the
SIP driver does for its control port.  This works, in the sense that my
driver is able to receive audio packets from my devices.  However, when I
queue up a voice frame via ast_queue_frame(), the frame doesn't appear to
make it through to the other side of the call (in this case, a SIP device).
I would prefer to use the read callback, if that's possible.

Any help will be appreciated.

Earle