hello every body,
i want to have h323 trunk between cisco 2800 and asterisk 11.13.1 with ooh323 module. i configured both side and have successful call from cisco to asterisk. but when call comes from asterisk to cisco, my phone rings but no audio is heard and call is disconnected after 5 second. i enable "debug voice rtp" in cisco and see the source address for receiving rtp packets is 0.0.0.0 Apr 28 07:46:34.765: RTP(50493): ps rx s=0.0.0.0(0), d=192.168.0.139(17112), pt=8, ts=BF40, ssrc=2C1690C9 any body knows how should i fix it? this is my ooh323.conf file: [general] port=1720 context=from-trunk gatekeeper=DISABLE bindaddr=192.X.X.X disallow=all allow=all AcceptAnonymous=yes directrtpsetup=yes directmedia=yes faststart=yes h245tunneling=yes mediawaitforconnect=yes tos=lowdelay [sam] type=user host=192.X.X.X directmedia=yes [sam-1] type=peer host=192.X.X.X directmedia=yes any comments or hints are really appreciated. SAM
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