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PAP2 won't make two g729 calls at the same time

I have a Linksys PAP2. Identical setups for the two channels in both the unit and in Asterisk. In particular, both channels enable g729 and set it as the preferred codec, and have disallow=all and allow=g729 in sip.conf.

If we make a call on one channel, it works (and uses g729), but if we make a call on the other channel when the first one is still connected, it fails. We have three g729 licenses, and no others were in use at the times this happened, but even if we didn't have enough, how would the PAP2 know that?

Here's a good, and a bad INVITE message, from the log file with sip debug enabled. Has anyone seen anything like this?

INVITE sip:59342 (at) 192.168.121.20 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.44:5060;branch=z9hG4bK-3e521aa
From: PAP 220 <sip:220 (at) 192.168.121.20>;tag=6b66e68deef168b2o0
To: <sip:59342 (at) 192.168.121.20>
Call-ID: 8e8903e9-18188b06 (at) 192.168.254.44
CSeq: 101 INVITE
Max-Forwards: 70
Contact: PAP 220 <sip:220 (at) 192.168.254.44:5060>
Expires: 240
User-Agent: Linksys/PAP2-3.1.3(LS)
Content-Length: 246
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 261305180 261305180 IN IP4 192.168.254.44
s=-
c=IN IP4 192.168.254.44
t=0 0
m=audio 16392 RTP/AVP 18 100 101
a=rtpmap:18 G729a/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

INVITE sip:203 (at) 192.168.121.20 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.44:5060;branch=z9hG4bK-3fda1e15
From: PAP 220 <sip:220 (at) 192.168.121.20>;tag=b8b86be991749af5o0
To: <sip:203 (at) 192.168.121.20>
Call-ID: a44265f9-c09c6825 (at) 192.168.254.44
CSeq: 101 INVITE
Max-Forwards: 70
Contact: PAP 220 <sip:220 (at) 192.168.254.44:5060>
Expires: 240
User-Agent: Linksys/PAP2-3.1.3(LS)
Content-Length: 267
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 261589835 261589835 IN IP4 192.168.254.44
s=-
c=IN IP4 192.168.254.44
t=0 0
m=audio 16400 RTP/AVP 0 8 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv




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