asterisk users - Dec 2003 - Native bridging with Polycom 600

Hi,

I cannot get two Polycom 600 phones to bridge natively. My sip.conf has 
canreinvite=yes for both phones. They connect, and I can talk as usual, but 
sniffing shows the RTP stream is routed through Asterisk.

The exact spot where the attempt to natively bridge fails is in rtp.c, line 
1281 (CVS from October 8, 2003):

f = ast_read(who);
if (!f || ((f->frametype == AST_FRAME_DTMF) &&
	(((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) ||
	((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)))))

A bit of logging shows the frame f is NULL, so Asterisk thinks one side has 
hung up, and gives up on the bridging attempt. Of course, the phones are both 
up.

Has anyone gotten these phones to bridge correctly, without the RTP stream 
traversing Asterisk? Do I need to update my CVS? I'd appreciate any advice.

Thanks,
Christian