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 <[EMAIL PROTECTED]>
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