[Serusers] Problem with sl_send_reply: I won't send a reply for ACK.

[Ricardo Martinez]

Hello list.
	I'm facing a problem with a UAC and i was hoping that someone can give me a hand here.
	I have a IP Phone calling to a PSTN number through SER and then a GW.

	10.0.0.243 : IP Phone
	10.0.0.246 : SER SIP Proxy
	10.0.0.239 : GW SIP PSTN

When the call is established and the "200 - OK" message arrives from the GW to the Proxy,  the proxy re-route the message back to the Client, and finally the client respond with an ACK.
Here is when te problem begins, i'm not sure if the ACK is the problem or maybe is a bug with my SER box.  I'm using the Getting Started ser.cfg from iptel.org.
You can see the debug here:

U 10.0.0.239:5060 -> 10.0.0.246:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 10.0.0.246;branch=z9hG4bK34e7.0648f244.0.
Via: SIP/2.0/UDP 10.0.0.243:5060;rport=5060;branch=z9hG4bK407006395.
From: <sip:5501234567 at sipvoiss.desa.mydomain.net>;tag=139103625.
To: <sip:0101005622408196 at sipvoiss.desa.mydomain.net>;tag=d745f073a4.
Call-ID: 90212623 at 10.0.0.243.
CSeq: 21 INVITE.
Supported: timer, replaces, early-session.
User-Agent: A SIP Gateway.
Contact: sip:005622408196 at 10.0.0.239.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO.
Content-Type: application/sdp.
Content-Length: 247.
Record-Route: <sip:10.0.0.246;ftag=139103625;lr=on>.
.
v=0.
o=005622408196 1170173661 1170173661 IN IP4 10.0.0.239.
s=A Gateway SDP.
c=IN IP4 10.0.0.239.
t=1170173661 0.
m=audio 23614 RTP/AVP 18 101.
a=rtpmap:18 G729/8000/1.
a=rtpmap:101 telephone-event/8000/1.
a=fmtp:101 0-15.
a=ptime:20.


U 10.0.0.246:5060 -> 10.0.0.243:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 10.0.0.243:5060;rport=5060;branch=z9hG4bK407006395.
From: <sip:5501234567 at sipvoiss.desa.mydomain.net>;tag=139103625.
To: <sip:0101005622408196 at sipvoiss.desa.mydomain.net>;tag=d745f073a4.
Call-ID: 90212623 at 10.0.0.243.
CSeq: 21 INVITE.
Supported: timer, replaces, early-session.
User-Agent: A SIP Gateway.
Contact: sip:005622408196 at 10.0.0.239.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO.
Content-Type: application/sdp.
Content-Length: 247.
Record-Route: <sip:10.0.0.246;ftag=139103625;lr=on>.
.
v=0.
o=005622408196 1170173661 1170173661 IN IP4 10.0.0.239.
s=A Gateway SDP.
c=IN IP4 10.0.0.239.
t=1170173661 0.
m=audio 23614 RTP/AVP 18 101.
a=rtpmap:18 G729/8000/1.
a=rtpmap:101 telephone-event/8000/1.
a=fmtp:101 0-15.
a=ptime:20.


U 10.0.0.243:5060 -> 10.0.0.246:5060
ACK sip:0101005622408196 at sipvoiss.desa.mydomain.net SIP/2.0.
Via: SIP/2.0/UDP 10.0.0.243:5060;rport;branch=z9hG4bK1227697472.
Route: <sip:10.0.0.246;ftag=139103625;lr=on>.
From: <sip:5501234567 at sipvoiss.desa.mydomain.net>;tag=139103625.
To: <sip:0101005622408196 at sipvoiss.desa.mydomain.net>;tag=d745f073a4.
Call-ID: 90212623 at 10.0.0.243.
CSeq: 21 ACK.
Contact: <sip:5501234567 at 10.0.0.243:5060>.
Max-Forwards: 70.
User-Agent: S SIP User Agent / 1.10.
Content-Length: 0.
.

Is this ACK ok?.  The ACK hits the "Sanity Checks" (Max Forwards) and then breaks sending to console : "Warning: sl_send_reply: I won't send a reply for ACK!!"
I was reading the RFC3261 because i'm not sure about the R-URI from this endpoint.  Let me explain :

RFC3261 : Section 12.1.2

   The route set MUST be set to the list of URIs in the Record-Route
   header field from the response, taken in reverse order and preserving
   all URI parameters.  If no Record-Route header field is present in
   the response, the route set MUST be set to the empty set.  This route
   set, even if empty, overrides any pre-existing route set for future
   requests in this dialog.  The remote target MUST be set to the URI
   from the Contact header field of the response.

So, if this is correct the R-URI from the ACk must be set to sip:005622408196 at 10.0.0.239, and not the sip:0101005622408196 at sipvoiss.desa.mydomain.net.
Is this ok?.
I think this is causing the ACK problem in my SER box.

Can someone help me here?
Thanks in advance.

Best Regards,
Ricardo Martinez.-

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