asterisk users - Apr 2007 - SIP<->H323 calls without proxying RTP

Hello,

Could somebody tell me is it possible to use asterisk without RTP proxying
in SIP<->H323 calls?
I mean exactly what canreinvite=yes option do in SIP<->SIP calls.
I don't need a transcoding, only a signaling conversion, and this is
possible with some softswitches, so i wondering what about asterisk.
Same question about H323<->H323 calls
I'm using NuFone Network's H323 cahhel

Thanks

--
Sincerely,
Elman Efendiyev
PROTECH INC.

On Fri, 27 Apr 2007, Elman Efendiyev said something to this effect:
> Hello,
>
> Could somebody tell me is it possible to use asterisk without RTP proxying
> in SIP<->H323 calls?
> I mean exactly what canreinvite=yes option do in SIP<->SIP calls.
> I don't need a transcoding, only a signaling conversion, and this is
> possible with some softswitches, so i wondering what about asterisk.
> Same question about H323<->H323 calls
> I'm using NuFone Network's H323 cahhel
   As an addendum to this, I would be curious to know how to force Asterisk 
to behave like a signaling proxy[1] only, if possible.  "CANreinvite" 
doesn't mean "WILLreinvite" or "MUSTreinvite."

-- Alex

[1]  Yes, I know it's a B2BUA so it's not really a proxy.  But the
intent
      here is to hand off the media path to the endpoints and not be
      involved in it.

--
Alex Balashov <sasha@presidium.org>