asterisk users - Feb 2005 - Pb DTMF with Asterisk vs Cirpack Transit Node

Hi all,

I've a problem in DTMF dialog between * and  a Cicpack Transit Node Class
4-5.
The call initiat by a mgcp phone pass by the cirpack and arrive in SIP on *, 
everything is ok (negociation and phone call) but when we try to use the 
voicemail, Asterisk don't understand DTMF.

Here are some logs (SIP debug on) on a DTMF '2' receive :

*************************************************************************
Sip read:
INFO sip:123@0.0.0.0:5060 SIP/2.0
Call-ID: 000000000000000000007ad91f05@cirpack
Contact: <sip:0.0.0.0:5060>
Content-Type: application/dtmf
CSeq: 2047555 INFO
From: "test003" 
<sip:0133333333@10.22.0.200;user=phone>;tag=000000000000000000007ad91f06
Max-Forwards: 31
To: <sip:123@0.0.0.0;user=phone>;tag=as2283a438
Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK-5764-36A3EE
Content-Length: 3

2

10 headers, 1 lines
Receiving DTMF!
Feb 28 08:27:44 WARNING[9223]: chan_sip.c:6116 receive_info: Unable to parse 
INFO message from 000000000000000000007ad91f05@cirpack. Content
Transmitting (NAT):
SIP/2.0 415 Unsupported media type
Via: SIP/2.0/UDP 
0.0.0.0:5060;branch=z9hG4bK-5764-36A3EE;received=0.0.0.0.0;rport=5060
From: "test003" 
<sip:0133333333@10.22.0.200;user=phone>;tag=000000000000000000007ad91f06
To: <sip:123@62.240.245.27;user=phone>;tag=as2283a438
Call-ID: 000000000000000000007ad91f05@cirpack
CSeq: 2047555 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:123@62.240.245.27>
Content-Length: 0
***************************************************************************

dtmfmode is RFC2833 in sip.conf


Thx in advance. Regards. Guy Decarpentrie.