asterisk users - Aug 2011 - Codec negotiation issue (no audio format found to offer)

Running build 1.8.5.0 (compiled from source) I seem to be having an issue
with codec negotiation. I have a Grandstream HT503 FXO port connected to a
pstn line, a Polycom SP501, and a SIP trunk with callwithus.

What I'm essentially looking to accomplish is for ulaw or g729 (preferably
ulaw) to be used to the Grandstream FXO or any other internal endpoint, and
for g729 only to be used outbound to my SIP trunk.

Here are the basics of my config, showing the codec list from "sip show
peer
<peer>":

Polycom SP501 (desk phone):
--------------------------
disallow=all
allow=ulaw&g729
  Codecs       : 0x104 (ulaw|g729)
  Codec Order  : (ulaw:20,g729:20)

Grandstream HT503 (fxo gateway):
--------------------------
disallow=all
allow=ulaw&g729
  Codecs       : 0x104 (ulaw|g729)
  Codec Order  : (ulaw:20,g729:20)

CallWithUs (SIP trunk):
--------------------------
disallow=all
allow=g729
  Codecs       : 0x100 (g729)
  Codec Order  : (g729:20)

When I place an outbound call from the Polycom to callwithus, the invite
from the pcom shows both ulaw and g729 in the SDP:
INVITE sip:************@192.168.0.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.201;branch=z9hG4bKc8aa981a8B8FF58D
From: "Office" <sip:2001 at 192.168.0.1>;tag=4CD2B2A0-B94A2531
To: <sip:919785013620 at 192.168.0.1;user=phone>
[...]
m=audio 2258 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000

Asterisk sees this:
[Aug  2 15:00:31] VERBOSE[1918] chan_sip.c: Capabilities: us - 0x104
(ulaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0x104 (ulaw|g729)

The call goes out the callwithus trunk:
[Aug  2 15:00:31] VERBOSE[1315] pbx.c:     -- Executing
[s at macro-dialout-trunk:19] Dial("SIP/2001-00000047",
"SIP/CallWithUs/**********,300,tTwW") in new stack

And then this, no INVITE goes out to callwithus at all:
[Aug  2 15:00:31] WARNING[1315] chan_sip.c: No audio format found to offer.
Cancelling call to **********
[Aug  2 15:00:31] VERBOSE[1315] app_dial.c:     -- Couldn't call
SIP/CallWithUs/**********

Similarly, if I set the Grandstream FXO trunk to ulaw only, the call fails
as well. It seems as if allowing only a single codec is the issue, if I
change the priorities of all codecs to g729 first and ulaw second, the call
goes through as g729 successfully.

Smells like a bug to me, but I may be overlooking something in my config.

Thanks,

-Ryan
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