G.723.1 rtp packetiztion problem


I am using Asterisk with meetme as a conference bridge.
VoIP client is using G.723.1 codec with rtp multipelexing - 4 G.723.1 frames are send in one UDP.
On server side I need also to send back the multiplexed frames, but it is not working.
The result, when caputuring the packectes, for one UDP send from VoIP client I get 4 packects from server.
I tried to set following options in sip.conf with no result:
In SDP packects send from server I see rtpmap:4 G.723/8000 fmtp:4 annexA=NO,ptime=120; while in SDP from client I see rtpmap:4 G.7231/8000 ftmtp:4 bitrate=6.3; annexA=YES.

Any advice/help appreciated.

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