Angelito Manansala wrote:
yes
On 11/21/05, Javier Oviedo <[EMAIL PROTECTED]> wrote:
Hi all,
for h323 to h323 can asterixk *proxy **call* with *signal **only* ( no rtp
go through asterisk )
thanks in advance
best regards!
Hello,
As far as I know Asterisk cannot disentangle RTP from signaling in
either SIP or H323 at least until now.
I'd also be interested to know if this option is available now in case
I've missed something...
Best regards,
Vlasis Hatzistavrou.
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