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hi all;
hi DANIEL;
I setup asterisk as a translator between sip-h323(I
used oh323 not native) . But there is a problem and it is as
follows:
when I try to dial FIRST from sip UA to h323
client , or h323 client to sip UA , it is ok
BUT the second try from any of
them to another have no audio.
any suggestion
Regards
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