One way audio after transfer the call

Hello, Everybody. I faced with next issue: sometimes after call transfer to another extension remote party lost audio channel (one way audio). It is looks like: someone call from outside (444777888) to asterisk. Call came on extension 101. A secretary pickup the call and transfer it to another extension 102 (attendant transfer, not blind), during the transfer remote party listen music on hold. Extensions 101 and 102 speaking normally. After 101 hang up, call transfered to 102. 102 say “Hello” but no one answer, although RTP channel is active in both directions. It is happened occasional, not every time, but quite often. We dumped traffic, investigated it and found that timestamp in the RTP flow sometimes has crazy values. It is very similar to the issue, which I have found on
The only point that it was asterisk version 1.4 but we run Seems that the problem was not fixed and what I suppose is that remote phone confuses by this crazy values of timestamps and break the call.
We use Dell PowerEdge R710 and kernel version is 2.6.18. Does anybody faced with such problem and could suggest why it may happens?