Short answer: yes.
Slightly longer: I have seen the same behavior. Without lr|lr=on the UA will (correctly) go over to strict routing and use contact in r-uri for the ACK. As long as only your record-route disappeared (i.e. you are the only hop in-between the UAS and UAC) this works.  As your external proxy is B2BUA (at least for signalling), your fine if you relay the ACK.
I'm not sure exactly when this happens and why, I seem to remember I posted something on this behavior for sjphone a while back, but I'm not capable of finding it.
g-)

Cesc wrote:
Hello everyone!

I have my system based on ser 0.9.6. Internal calls work just fine.
I am now trying to interop with another system. It has a sort of
asterisk functionality, but it is not asterisk, it is a private
software company product.

my.phone --------->   my.ser.proxy  ------------->   external.proxy
------------->  external phone
10.111.0.119           10.111.0.50                   10.111.0.20
                10.111.0.144

There is no firewall or nat in the way.

The problem, i think, is that the external.proxy is buggy. I told
that to the company, but who knows when they will fix this.

* Look at the OK (message #9 and #10). My.ser.proxy record-routes all
invites. The Record-route headers reach the external.phone, which
copies them in the OK
message, sends the OK to the external.proxy ... and when this forwards
it to my.ser.proxy, they are gone! The OK reaches my.phone, but then
it generates my
problem: the ACK. It contains NO ROUTE headers and the r-uri is also
simply pointing to the external.proxy.
If the ACK had the ROUTE headers, my.ser.proxy would loose_route the
message and voila!
But as loose_route() returns false, my (maybe bad) config file gets
confused and treats it like a "new" call ... so it lookup("location")
of the ACK r-ruri fails,
and the ACK is dropped.
Should i modify the config file so that ACK, if not loose_route'd, are
simply t_relay'd?

* They also modify the contact field. The reason is because behind the
external.proxy could be H323 or SIP phones, so they sort of want to be
a termination as far as signalling is concerned.

I attach the message flow, hope it helps ... Thanks!

Cesc

No. Time Source Destination Protocol Info 1 0.000000000 10.111.0.119 10.111.0.50 SIP/SDP Request: INVITE sip:6007@10.111.0.20, with session description Session Initiation Protocol Request-Line: INVITE sip:6007@10.111.0.20 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.111.0.119;rport;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e Content-Length: 337 Contact: <sip:7005@10.111.0.119:5060> Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 Content-Type: application/sdp CSeq: 1 INVITE From: "7005"<sip:7005@10.111.0.50:5060>;tag=12083790462495 Max-Forwards: 70 To: <sip:6007@10.111.0.20> User-Agent: SJphone/1.60.289a (SJ Labs) Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3369481881 3369481881 IN IP4 10.111.0.119 Session Name (s): SJphone Connection Information (c): IN IP4 10.111.0.119 Time Description, active time (t): 0 0 Session Attribute (a): direction:active Media Description, name and address (m): audio 49248 RTP/AVP 3 97 98 8 0 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:97 iLBC/8000 Media Attribute (a): rtpmap:98 iLBC/8000 Media Attribute (a): fmtp:98 mode=20 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16 No. Time Source Destination Protocol Info 3 0.001209000 10.111.0.50 10.111.0.20 SIP/SDP Request: INVITE sip:6007@10.111.0.20:5060, with session description Session Initiation Protocol Request-Line: INVITE sip:6007@10.111.0.20:5060 SIP/2.0 Message Header Record-Route: <sip:10.111.0.50;ftag=12083790462495;lr=on> Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0 Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e Content-Length: 337 Contact: <sip:7005@10.111.0.119:5060> Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 Content-Type: application/sdp CSeq: 1 INVITE From: "7005"<sip:7005@10.111.0.50:5060>;tag=12083790462495 Max-Forwards: 16 To: <sip:6007@10.111.0.20> User-Agent: SJphone/1.60.289a (SJ Labs) Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3369481881 3369481881 IN IP4 10.111.0.119 Session Name (s): SJphone Connection Information (c): IN IP4 10.111.0.119 Time Description, active time (t): 0 0 Session Attribute (a): direction:active Media Description, name and address (m): audio 49248 RTP/AVP 3 97 98 8 0 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:97 iLBC/8000 Media Attribute (a): rtpmap:98 iLBC/8000 Media Attribute (a): fmtp:98 mode=20 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16 No. Time Source Destination Protocol Info 5 0.040629000 10.111.0.20 10.111.0.144 SIP/SDP Request: INVITE sip:6007@10.111.0.144:5060;transport=UDP, with session description Session Initiation Protocol Request-Line: INVITE sip:6007@10.111.0.144:5060;transport=UDP SIP/2.0 Message Header Via: SIP/2.0/UDP 10.111.0.20:5060;branch=z9hG4bKm27749469 Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0 Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e RECORD-ROUTE: <sip:10.111.0.50;ftag=12083790462495;lr=on> From: "7005"<sip:7005@10.111.0.50:5060>;tag=12083790462495 To: <sip:6007@10.111.0.20> Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 INVITE Max-Forwards: 16 Contact: <sip:10.111.0.20> User-Agent: SJphone/1.60.289a (SJ Labs) Content-Type: application/sdp Content-Length: 337 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3369481881 3369481881 IN IP4 10.111.0.119 Session Name (s): SJphone Connection Information (c): IN IP4 10.111.0.119 Time Description, active time (t): 0 0 Session Attribute (a): direction:active Media Description, name and address (m): audio 49248 RTP/AVP 3 97 98 8 0 101 Media Attribute (a): rtpmap:3 gsm/8000 Media Attribute (a): rtpmap:97 ilbc/8000 Media Attribute (a): rtpmap:98 ilbc/8000 Media Attribute (a): fmtp:98 mode=20 Media Attribute (a): rtpmap:8 pcma/8000 Media Attribute (a): rtpmap:0 pcmu/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16 No. Time Source Destination Protocol Info 9 3.736665000 10.111.0.144 10.111.0.20 SIP/SDP Status: 200 OK, with session description Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.111.0.20:5060;branch=z9hG4bKm27749469 Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0 Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e Record-Route: <sip:10.111.0.50;ftag=12083790462495;lr=on> Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 INVITE From: "7005"<sip:7005@10.111.0.50:5060>;tag=12083790462495 To: <sip:6007@10.111.0.20>;tag=0rxFXxkiza46soF7 Contact: <sip:6007@10.111.0.144:5060> Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, NOTIFY, INFO, PRACK, UPDATE Supported: 100rel, replaces Content-Type: application/sdp Content-Length: 140 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 6007 14194531 23224216 IN IP4 10.111.0.144 Session Name (s): SIP CALL Connection Information (c): IN IP4 10.111.0.144 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 10000 RTP/AVP 8 Media Attribute (a): rtpmap:8 PCMA/8000 No. Time Source Destination Protocol Info 10 3.751306000 10.111.0.20 10.111.0.50 SIP/SDP Status: 200 OK, with session description Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0 Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e From: "7005"<sip:7005@10.111.0.50:5060>;tag=12083790462495 To: <sip:6007@10.111.0.20>;tag=0rxFXxkiza46soF7 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 INVITE Contact: <sip:10.111.0.20:5060> Content-Type: application/sdp Content-Length: 140 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 6007 14194531 23224216 IN IP4 10.111.0.144 Session Name (s): SIP CALL Connection Information (c): IN IP4 10.111.0.144 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 10000 RTP/AVP 8 Media Attribute (a): rtpmap:8 pcma/8000 No. Time Source Destination Protocol Info 11 3.751589000 10.111.0.50 10.111.0.119 SIP/SDP Status: 200 OK, with session description Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e From: "7005"<sip:7005@10.111.0.50:5060>;tag=12083790462495 To: <sip:6007@10.111.0.20>;tag=0rxFXxkiza46soF7 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 INVITE Contact: <sip:10.111.0.20:5060> Content-Type: application/sdp Content-Length: 140 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 6007 14194531 23224216 IN IP4 10.111.0.144 Session Name (s): SIP CALL Connection Information (c): IN IP4 10.111.0.144 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 10000 RTP/AVP 8 Media Attribute (a): rtpmap:8 pcma/8000 No. Time Source Destination Protocol Info 12 3.756127000 10.111.0.119 10.111.0.50 SIP Request: ACK sip:10.111.0.20:5060 Session Initiation Protocol Request-Line: ACK sip:10.111.0.20:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.111.0.119;rport;branch=z9hG4bK0a6f0077000004b2452bb81d00000c5f00001922 Content-Length: 0 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 ACK From: "7005"<sip:7005@10.111.0.50:5060>;tag=12083790462495 Max-Forwards: 70 To: <sip:6007@10.111.0.20>;tag=0rxFXxkiza46soF7 User-Agent: SJphone/1.60.289a (SJ Labs)

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