see http://www.voip-info.org/tiki-index.php?page=Bug+3986+Bounty
about a $1k bounty for fixing bug 3986
Just uploaded the fix.
It does clean apply to asterisk 1.0.7.
There's no reason to keep the RTP media open (in case of
sip_destroy failure) so the patch destroy it in the _hangup request.
It should let you use all the available RTP ports.
unfortunately, this didn't fix the bug
roy
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