asterisk users - Jul 2004 - SIP - Cancel request fails with "481 no such call"

Hi,

I am using SIP extensions connected to the PSTN with the CAPI Channel
driver.
All works fine except that one of the sip phones keeps ringing when the
caller
hangs up before extension is answered. The phones are grandstream 100,
though
we get the same behaviour using other phones (X-lite, Kphone).
It behaves the same regardless of whether the incoming call is from a SIP
extension or an external PSTN call through the CAPI Channel.
The SIP debug (see below) shows that the response to the SIP "Cancel"
request
is "481 no such call". The call-ID seems to match the corresponding
invite.

I suspect it has something to do with the NAT setup:-

phone1__
                |
                |--NAT---internet---NAT---phone2
asterisk__|

Calls to phone1 work fine but to phone2 I get the 481 response to a cancel.
The SIP Debug is from an external call to the SIP extension (Phone2), so the
debug output is for the call from Asterisk to the called extension.

I'm currently using the latest stable (17th July) but have also tried it
with CVS head from 7th July.

Any help welcome.
Or am I lucky to get this much working through the 2 NAT devices?


Paul



    -- started pbx on channel (callgroup=2)!
    -- Executing Dial("CAPI[contr1/13]/13",
"SIP/alan&SIP/21|15") in new
stack

We're at 192.168.1.201 port 13854
Answering with preferred capability 1024
Answering with non-codec capability 1
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:21@80.229.52.129 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK69a5a54e
From: "asterisk" <sip:asterisk@192.168.1.201>;tag=as615a71fa
To: <sip:21@80.229.52.129>
Contact: <sip:asterisk@192.168.1.201>
Call-ID: 6bb9546b48c690322bf516e754566fe8@192.168.1.201
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 23 Jul 2004 09:58:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 220

v=0
o=root 30736 30736 IN IP4 192.168.1.201
s=session
c=IN IP4 192.168.1.201
t=0 0
m=audio 13854 RTP/AVP 97 101
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 (NAT) to 80.229.52.129:5060
    -- Called 21


Sip read:
SIP/2.0 100 trying
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK69a5a54e
From: "asterisk" <sip:asterisk@192.168.1.201>;tag=as615a71fa
To: <sip:21@80.229.52.129>
Call-ID: 6bb9546b48c690322bf516e754566fe8@192.168.1.201
CSeq: 102 INVITE
User-Agent: Grandstream BT100 1.0.5.8
Content-Length: 0


8 headers, 0 lines


Sip read:
SIP/2.0 180 ringing
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK69a5a54e
From: "asterisk" <sip:asterisk@192.168.1.201>;tag=as615a71fa
To: <sip:21@80.229.52.129>;tag=7414989cd298e1ab
Call-ID: 6bb9546b48c690322bf516e754566fe8@192.168.1.201
CSeq: 102 INVITE
User-Agent: Grandstream BT100 1.0.5.8
Content-Length: 0



8 headers, 0 lines
    -- SIP/21-8663 is ringing
Reliably Transmitting:
CANCEL sip:21@80.229.52.129 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK69a5a54e
From: "asterisk" <sip:asterisk@192.168.1.201>;tag=as615a71fa
To: <sip:21@80.229.52.129>
Contact: <sip:asterisk@192.168.1.201>
Call-ID: 6bb9546b48c690322bf516e754566fe8@192.168.1.201
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

 (NAT) to 80.229.52.129:5060
  == Spawn extension (isdndefault, 13, 1) exited non-zero on
'CAPI[contr1/13]/13'


Sip read:
SIP/2.0 481 no such call
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK69a5a54e
From: "asterisk" <sip:asterisk@192.168.1.201>;tag=as615a71fa
To: <sip:21@80.229.52.129>;tag=c7c5e72751d90dc7
Call-ID: 6bb9546b48c690322bf516e754566fe8@192.168.1.201
CSeq: 102 CANCEL
User-Agent: Grandstream BT100 1.0.5.8
Content-Length: 0


8 headers, 0 lines
    -- Got SIP response 481 "no such call" back from 80.229.52.129