more detailed log from asterisk:

Found description format X-CCD
Found description format telephone-event
Found description format CN
Capabilities: us - 0x1f07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h263p), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10f (g723|gsm|ulaw|alaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event)
Looking for 08281895109 in default (domain 194.247.167.90)
list_route: hop: <sip:195.62.225.244;ftag=6D3877B8-22D9;lr=on>
Transmitting (no NAT) to 195.62.225.244:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 195.62.225.244;branch=z9hG4bKbd65.9a7b14c.0;received=195.62.225.244
Via: SIP/2.0/UDP 83.211.2.132:5060;branch=z9hG4bK154DC2AC0
From: "anonymous" <sip:83.211.2.132>;tag=6D3877B8-22D9
To: <sip:08281895109@voip.eutelia.it>
Call-ID: D45478BC-CF0711DA-9B28AEF4-CD9DEADE@83.211.2.132
CSeq: 101 INVITE
User-Agent: Convergenze VoGW1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:08281895109@194.247.167.90>
Content-Length: 0

---
Transmitting (no NAT) to 195.62.225.244:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 195.62.225.244;branch=z9hG4bKbd65.9a7b14c.0;received=195.62.225.244
Via: SIP/2.0/UDP 83.211.2.132:5060;branch=z9hG4bK154DC2AC0
From: "anonymous" <sip:83.211.2.132>;tag=6D3877B8-22D9
To: <sip:08281895109@voip.eutelia.it>;tag=as51e6fe91
Call-ID: D45478BC-CF0711DA-9B28AEF4-CD9DEADE@83.211.2.132
CSeq: 101 INVITE
User-Agent: Convergenze VoGW1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:08281895109@194.247.167.90>
Content-Length: 0

---
We're at 194.247.167.90 port 16100
Adding codec 0x1 (g723) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 195.62.225.244:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.62.225.244;branch=z9hG4bKbd65.9a7b14c.0;received=195.62.225.244
Via: SIP/2.0/UDP 83.211.2.132:5060;branch=z9hG4bK154DC2AC0
Record-Route: <sip:195.62.225.244;ftag=6D3877B8-22D9;lr=on>
From: "anonymous" <sip:83.211.2.132>;tag=6D3877B8-22D9
To: <sip:08281895109@voip.eutelia.it>;tag=as51e6fe91
Call-ID: D45478BC-CF0711DA-9B28AEF4-CD9DEADE@83.211.2.132
CSeq: 101 INVITE
User-Agent: Convergenze VoGW1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:08281895109@194.247.167.90>
Content-Type: application/sdp
Content-Length: 494

v=0
o=root 5557 5557 IN IP4 194.247.167.90
s=session
c=IN IP4 194.247.167.90
t=0 0
m=audio 16100 RTP/AVP 4 3 0 8 111 5 10 7 18 110 97 101
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
voipgw1*CLI>
<-- SIP read from 195.62.225.244:5060:
ACK sip:08281895109@194.247.167.90:5060 SIP/2.0
Record-Route: <sip:195.62.225.244;ftag=6D3877B8-22D9;lr=on>
Via: SIP/2.0/UDP 195.62.225.244;branch=0
Via: SIP/2.0/UDP 83.211.2.132:5060;branch=z9hG4bK154DCE1139
From: "anonymous" <sip:83.211.2.132>;tag=6D3877B8-22D9
To: <sip:08281895109@voip.eutelia.it>;tag=as51e6fe91
Date: Wed, 19 Apr 2006 18:19:09 GMT
Call-ID: D45478BC-CF0711DA-9B28AEF4-CD9DEADE@83.211.2.132
Max-Forwards: 9
CSeq: 101 ACK
Content-Length: 0
P-hint: rr-enforced

--- (12 headers 0 lines)---
set_destination: Parsing <sip:195.62.225.244;ftag=6D3877B8-22D9;lr=on> for address/port to send to
set_destination: set destination to 195.62.225.244, port 5060
We're at 194.247.167.90 port 16100
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
14 headers, 11 lines
Reliably Transmitting (no NAT) to 195.62.225.244:5060:
INVITE sip:83.211.2.132:5060 SIP/2.0
Via: SIP/2.0/UDP 194.247.167.90:5060;branch=z9hG4bK0ef3d8d2;rport
Route: <sip:195.62.225.244;ftag=6D3877B8-22D9;lr=on>
From: <sip:08281895109@voip.eutelia.it>;tag=as51e6fe91
To: "anonymous" <sip:83.211.2.132>;tag=6D3877B8-22D9
Contact: <sip:08281895109@194.247.167.90>
Call-ID: D45478BC-CF0711DA-9B28AEF4-CD9DEADE@83.211.2.132
CSeq: 102 INVITE
User-Agent: Convergenze VoGW1
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 241

v=0
o=root 5557 5558 IN IP4 194.247.167.90
s=session
c=IN IP4 194.247.167.90
t=0 0
m=audio 35226 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
voipgw1*CLI>
<-- SIP read from 195.62.225.244:5060:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 194.247.167.90:5060;branch=z9hG4bK0ef3d8d2;rport=5060
From: <sip:08281895109@voip.eutelia.it>;tag=as51e6fe91
To: "anonymous" <sip:83.211.2.132>;tag=6D3877B8-22D9
Call-ID: D45478BC-CF0711DA-9B28AEF4-CD9DEADE@83.211.2.132
CSeq: 102 INVITE
Server: SPS01EUT(0.9.6 (i386/linux))
Content-Length: 0
Warning: 392 195.62.225.244:5060 "Noisy feedback tells: pid=816 req_src_ip=194.247.167.90 req_src_port=5060 in_uri=sip:83.211.2.132:5060 out_uri=sip:83.211.2.132:5060 via_cnt==1"

--- (9 headers 0 lines)---
voipgw1*CLI>
<-- SIP read from 195.62.225.244:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 194.247.167.90:5060;branch=z9hG4bK0ef3d8d2;rport=5060
From: <sip:08281895109@voip.eutelia.it>;tag=as51e6fe91
To: "anonymous" <sip:83.211.2.132>;tag=6D3877B8-22D9
Date: Wed, 19 Apr 2006 18:19:19 GMT
Call-ID: D45478BC-CF0711DA-9B28AEF4-CD9DEADE@83.211.2.132
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: <sip:83.211.2.132:5060>
Record-Route: <sip:195.62.225.244;ftag=as51e6fe91;lr=on>
Content-Type: application/sdp
Content-Length: 279

v=0
o=CiscoSystemsSIP-GW-UserAgent 3839 2702 IN IP4 83.211.2.132
s=SIP Call
c=IN IP4 83.211.2.133
t=0 0
m=audio 16682 RTP/AVP 18 101
c=IN IP4 83.211.2.133
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=direction:passive

--- (14 headers 12 lines)---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 83.211.2.133:16682
Found description format G729
Found description format telephone-event
Capabilities: us - 0x1f07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h263p), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Transmitting (no NAT) to 195.62.225.244:5060:
ACK sip:83.211.2.132:5060 SIP/2.0
Via: SIP/2.0/UDP 194.247.167.90:5060;branch=z9hG4bK3f823695;rport
Route: <sip:195.62.225.244;ftag=as51e6fe91;lr=on>
From: <sip:08281895109@voip.eutelia.it>;tag=as51e6fe91
To: "anonymous" <sip:83.211.2.132>;tag=6D3877B8-22D9
Contact: <sip:08281895109@194.247.167.90>
Call-ID: D45478BC-CF0711DA-9B28AEF4-CD9DEADE@83.211.2.132
CSeq: 102 ACK
User-Agent: Convergenze VoGW1
Max-Forwards: 70
Content-Length: 0

---
set_destination: Parsing <sip:195.62.225.244;ftag=as51e6fe91;lr=on> for address/port to send to
set_destination: set destination to 195.62.225.244, port 5060
Reliably Transmitting (no NAT) to 195.62.225.244:5060:
BYE sip:83.211.2.132:5060 SIP/2.0
Via: SIP/2.0/UDP 194.247.167.90:5060;branch=z9hG4bK42a48a40;rport
Route: <sip:195.62.225.244;ftag=as51e6fe91;lr=on>
From: <sip:08281895109@voip.eutelia.it>;tag=as51e6fe91
To: "anonymous" <sip:83.211.2.132>;tag=6D3877B8-22D9
Contact: <sip:08281895109@194.247.167.90>
Call-ID: D45478BC-CF0711DA-9B28AEF4-CD9DEADE@83.211.2.132
CSeq: 103 BYE
User-Agent: Convergenze VoGW1
Max-Forwards: 70
Content-Length: 0

---
voipgw1*CLI>
<-- SIP read from 195.62.225.244:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 194.247.167.90:5060;branch=z9hG4bK42a48a40;rport=5060
From: <sip:08281895109@voip.eutelia.it>;tag=as51e6fe91
To: "anonymous" <sip:83.211.2.132>;tag=6D3877B8-22D9
Date: Wed, 19 Apr 2006 18:19:19 GMT
Call-ID: D45478BC-CF0711DA-9B28AEF4-CD9DEADE@83.211.2.132
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 103 BYE

THANSK

----- Original Message -----

From: Rosario Pingaro

To: serusers@lists.iptel.org

Sent: Wednesday, April 19, 2006 1:49 PM

Subject: [Norton AntiSpam] [Serusers] Fw: 400 Bad Request after an ACK

Can someone help me to debug my problem?

I have ser between asterisk and my clients.

When I try to call a sip client, it is going to ring. But asap the callee pickup the phone the call goes down.

Doing a logging on port 5060 i see that after the ack i get 400 bad request from the sip client.

This is the trace:

Session Initiation Protocol
    Request-Line: ACK sip:0681140017@83.211.248.158:62746 SIP/2.0
        Method: ACK
        Resent Packet: False
    Message Header
        Record-Route: <sip:19x.6x.19x.4x;ftag=as6d07dd0a;lr=on>
        Via: SIP/2.0/UDP 19x.6x.19x.4x;branch=0
        Via: SIP/2.0/UDP 19x.24x.16x.9x:5060;branch=z9hG4bK368ed369;rport=5060
        From: "anonymous" <sip:asterisk@voip.convergenze.it>;tag=as6d07dd0a
            SIP Display info: "anonymous"
            SIP from address: sip:asterisk@voip.convergenze.it
            SIP tag: as6d07dd0a
        To: <sip:08281895109@voip.convergenze.it>;tag=b1385811e50f0aai1
            SIP to address: sip:08281895109@voip.convergenze.it
            SIP tag: b1385811e50f0aai1
        Contact: <sip:asterisk@19x.24x.16x.9x>
            Contact Binding: <sip:asterisk@19x.24x.16x.9x>
                URI: <sip:asterisk@19x.24x.16x.9x>
                    SIP contact address: sip:asterisk@19x.24x.16x.9x
        Call-ID: 3ff0ae307575f1df61408b205af01196@voip.convergenze.it
        CSeq: 102 ACK
        User-Agent: Convergenze VoGW1
        Max-Forwards: 16
        Content-Length: 0

and then

Session Initiation Protocol
    Status-Line: SIP/2.0 400 Bad Request
        Status-Code: 400
        Resent Packet: False
    Message Header
        To: <sip:08281895109@voip.convergenze.it>;tag=b1385811e50f0aai1
            SIP to address: sip:08281895109@voip.convergenze.it
            SIP tag: b1385811e50f0aai1
        From: "anonymous" <sip:asterisk@voip.convergenze.it>;tag=as6d07dd0a
            SIP Display info: "anonymous"
            SIP from address: sip:asterisk@voip.convergenze.it
            SIP tag: as6d07dd0a
        Call-ID: 3ff0ae307575f1df61408b205af01196@voip.convergenze.it
        CSeq: 103 INVITE
        Via: SIP/2.0/UDP 19x.6x.19x.4x;branch=z9hG4bK9f79.7854d9e5.0
        Via: SIP/2.0/UDP 19x.24x.16x.9x:5060;branch=z9hG4bK493c6be2;rport=5060
        Record-Route: <sip:19x.6x.19x.4x;ftag=as6d07dd0a;lr=on>
        Server: Sipura/SPA2100-3.2.5(d)
        Content-Length: 0

Any help is appreciated.

Regards

Rosairio

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