[Asterisk-Users] Testing H323

[VSM-Hosting]

Hello

Can or is there somewhere a way to test my outgoing H323
I like to connect to a terminating server but I'm still getting hangups. 
Phone is ringing on the othersite but my asterisk telling my no one 
availble at this moment.

Like to test my H323 loutgoing line.
I't looks so stuppid if something wrong on my site.
Using Nufone H323 compiled pwlib and openH323 correct..
My error messsages.

-- Executing Dial("SIP/4786042-aa0e", "H323/XXXXXXXX at 69.XX.XX.XX") in 
new stack
-- Called XXXXXXXX at 69.XX.XX.XX
-- H323/69.XX.XX.XX is ringing
== No one is available to answer at this time
-- H323/69.XX.XX.XX answered SIP/142-aa0e
Nov 11 19:50:54 NOTICE[1144222912]: rtp.c:289 process_rfc3389: RFC3389 
support incomplete. Turn off on client if possible
== Spawn extension (terminator, XXXXXXXX, 1) exited non-zero on 
'SIP/142-aa0e'



Thanks


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