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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040718/9390bbdb/attachment.htm