[SPAM] Re: [asterisk-dev] RTP streams between H323 and SIP

[Pavel Jezek]

very interesting, so it means that this "bridge code" is currently in 
asterisk, or you have some patch for this?
I would like to test this! :-)
PJ


Paul Cadach wrote:
> I just acknowledge we have H.323 native bridge code that support RTP "move" (like re-invites for SIP) and all works fine
> between SIP and H.323 channels.
>
>
> WBR,
> Paul.
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