asterisk users - Sep 2007 - Asterisk 1.4 Ignoring SIP ACK's on 487 Responses

Hi,

I've been doing some testing on moving from 1.2 to 1.4 and one issue
I've encountered is re-transmits whenever an INVITE is cancelled. I have a
stateless SIP proxy in fron of my asterisk servers (all it does is direct
requests to one asteisk server or another) and the re-transmits do not occur on
1.2.17 which is the current verion I have in use on my production servers.

The retransmits do not occur on a 200 Ok Response. When the INVITE is cancelled
the CANCEL request is acted on correctly and the cll is cancelled and the only
problem is the 6 retransmits of the INVITE response everytime a call is
cancelled. I've confirmed that the ACK request is getting through to the
Asterisk 1.4 server and also checked that all the required transaction fields in
the ACK are correct (Call-Id, From and Via branch of original INVITE and To of
response). I've also checked with two different user agents (Bria softphone
and Polycom IP300) and both exhibit the same problem.

Below is a trace of the relevant SIP messages. I'm wondering if Asterisk is
not coping with the Contact header not being present although no message is
coming up on the console to that effect.

Regards,

Greyman.


INVITE sip:1234 at server SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKbfe545ae83ef7d0ec9fe44ea063d72c67f4bc926
Via: SIP/2.0/UDP
192.168.1.102:4110;rport=10260;branch=z9hG4bK-d87543-302f8c6313727f46-1--d87543-
To: <sip:1234 at server>
From: <sip:user at server>;tag=6e2df459
Call-ID: ZTFmYjU1OWVhZDFlOTMxN2NlM2NhNzdlYzBmNjZiNWI.
CSeq: 2 INVITE
Contact: <sip:user at 192.168.1.102>
Max-Forwards: 69
Record-Route: <sip:10.0.0.1;lr>
User-Agent: Bria release 2.0 stamp 40829
Content-Type: application/sdp
Content-Length: 616
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
....

Retransmitting #4 (NAT) to 10.0.0.1:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP
10.0.0.1;branch=z9hG4bKbfe545ae83ef7d0ec9fe44ea063d72c67f4bc926;received=194.213.29.100
Via: SIP/2.0/UDP
192.168.1.102:4110;rport=10260;branch=z9hG4bK-d87543-302f8c6313727f46-1--d87543-
From: <sip:user at server>;tag=6e2df459
To: <sip:1234 at server>;tag=as3a3770b5
Call-ID: ZTFmYjU1OWVhZDFlOTMxN2NlM2NhNzdlYzBmNjZiNWI.
CSeq: 2 INVITE
User-Agent: asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<--- SIP read from 10.0.0.1 --->
ACK sip:1234 at server SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bK13da7f631fc65ea06570ce13f322cc7b142074a2
Via: SIP/2.0/UDP
192.168.1.102:4110;rport=10260;branch=z9hG4bK-d87543-302f8c6313727f46-1--d87543-
To: <sip:1234 at server>;tag=as3a3770b5
From: <sip:user at server>;tag=6e2df459
Call-ID: ZTFmYjU1OWVhZDFlOTMxN2NlM2NhNzdlYzBmNjZiNWI.
CSeq: 2 ACK




     
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