recording a SIP-telephony conversation

Hello everybody,

today I learned about how powerful JACK is, and did my first
steps
inside qjackctl. Thanks a lot for writing and maintaining
this fine
system!

I found a problem and maybe one of you could just take a
glance and it
and tell me *who* might be guilty. My case is not too
serious - I'm
using a different application right now, and everything
seems to work.
I'm just curious what's going wrong here and who is to
blame! 

I want to record SIP-telephone conversations. I tried
"twinkle" (I want
an open-source application where I can type in my own SIP
provider
data). Running Debian unstable, twinkle 0.8.1, jackd
0.101.1, ALSA
1.0.12, kernel 2.6.13.2 on i386. Ah, and of course plain
twinkle with
direct ALSA access works 

After figuring out how I can put the JACK Timemachine into
qjackctl, of
course I wanted to put twinkle inside the JACK system as
well, and I
tried (1) jacklaunch, (2) artsdsp and (3) a
"virtual" ALSA device
connected to JACK.

(1) jacklaunch (twinkle configured to use OSS)

Incoming call is signaled through the speakers, but when
both playback
and capture device should be opened for the call, the
application
crashes and on the console I read:

unknown ioctl
unknown ioctl
unknown ioctl
unknown ioctl
unknown ioctl
ioctl: SNDCTL_DSP_SETDUPLEX not implemented

Does this indicate that my application wants something
jacklaunch can't
deliver, or might it be that my kernel's ALSA is too old?
(kernel
2.6.13.2)

(2) artsdsp (twinkle with OSS, artsd connected to jackd)

- I know that there is a lot of additional latency through
this, but I
tried it because some people got it successfully working
with other
(non-free) telephony software.

If I just use twinkle and nothing else, it sometimes works
perfectly,
both ends can hear each other. However, when I try to attach
also the
JACK Timemachine to the microphone output (which is
alsa_pcm:capture1),
"artsdsp twinkle" doesn't get any microphone
input any more (the
Timemachine recorder gets it), both look connected in
qjackctl.

(3) ALSA "default" configured to use JACK
as described on e.g. http://alsa.opensr
c.org/JACKPlugin

Again, like (1), ringing works, but when opening playback
and microphone
device for the real call, I see new devices popping up in
qjackctl, but
I'm stuck in a connect/disconnect loop, quite funny, the
button
"Disconnect All" flashes, and on the console I
get LOTS of identical
lines:

cannot connect alsa-jack.jackP.10117.3:out_000 to
alsa_pcm:playback_1

What could that be? An error in JACK? or rather my twinkle
application?
Please say if I shall deliver any additional information.

Thank you for reading,
have a nice day,
Viktor.


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