Thanks, I appreciate it.

In this setup the there are 2 endpoints (700 and 701) peered up to an Asterisk server (172.16.60.6) via a Kamailio proxy (172.16.60.20).  700 (172.16.60.28) is calling 701 (172.16.3.65).  When 701 answers the OK is sent to the proxy and then to Asterisk.  Asterisk is then ACKing the OK.  The ACK is being sent to the proxy and then the proxy should be sending it back to the endpoint.  It is not.  The ACK is being sent to the proxy and then the proxy is sending to itself again, via the loopback interface.  I believe loose_route() should be re-writing the destination to be the endpoint, but it not.

Trace:

U 172.16.60.28:54936 -> 172.16.60.20:5060
INVITE sip:701@eng-reg1.example.com SIP/2.0.
Via: SIP/2.0/UDP 172.16.60.28:54936;rport;branch=z9hG4bKPjBsmwQX-oWu6srxf09JRx98W5IklAwE44.
Max-Forwards: 70.
From: <sip:sip700_tbs@eng-reg1.example.com>;tag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P.
To: <sip:701@eng-reg1.example.com>.
Contact: <sip:sip700_tbs@172.16.60.28:54936;ob>.
Call-ID: BUDvyYQczBQPRIOhj7t7FodP7675QPGi.
CSeq: 13862 INVITE.
Route: <sip:eng-reg1.example.com;transport=udp;lr>.
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS.
Supported: replaces, 100rel, timer, norefersub.
Session-Expires: 1800.
Min-SE: 90.
User-Agent: CSipSimple_mecha-15/r2272.
Proxy-Authorization: Digest username="sip700_tbs", realm="sip700_tbs", nonce="Uh39eVId/E2Vxz5hgWC/7jMNGAf7rxrV", uri="sip:701@eng-reg1.example.com", response="8a74a8727baa45df84ea1374cb6668f2", cnonce="EYGYpa1zWmEXcOMighUzGZ20cY2HJ7AJ", qop=auth, nc=00000001.
Content-Type: application/sdp.
Content-Length:   340.
.
v=0.
o=- 3586685645 3586685645 IN IP4 172.16.60.28.
s=pjmedia.
c=IN IP4 172.16.60.28.
t=0 0.
m=audio 4004 RTP/AVP 99 0 8 101.
c=IN IP4 172.16.60.28.
a=rtcp:4005 IN IP4 172.16.60.28.
a=sendrecv.
a=rtpmap:99 SILK/24000.
a=fmtp:99 useinbandfec=0.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.


U 172.16.60.20:5060 -> 172.16.60.28:54936
SIP/2.0 100 trying -- your call is important to us.
Via: SIP/2.0/UDP 172.16.60.28:54936;rport=54936;branch=z9hG4bKPjBsmwQX-oWu6srxf09JRx98W5IklAwE44.
From: <sip:sip700_tbs@eng-reg1.example.com>;tag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P.
To: <sip:701@eng-reg1.example.com>.
Call-ID: BUDvyYQczBQPRIOhj7t7FodP7675QPGi.
CSeq: 13862 INVITE.
Server: kamailio (4.0.3 (x86_64/linux)).
Content-Length: 0.
.


U 172.16.60.20:5060 -> 172.16.60.6:5060
INVITE sip:701@eng-reg1.example.com SIP/2.0.
Record-Route: <sip:172.16.60.20;lr=on;ftag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P;nat=yes>.
Via: SIP/2.0/UDP 172.16.60.20;branch=z9hG4bKe51f.d2338795.0.
Via: SIP/2.0/UDP 172.16.60.28:54936;rport=54936;branch=z9hG4bKPjBsmwQX-oWu6srxf09JRx98W5IklAwE44.
Max-Forwards: 16.
From: <sip:sip700_tbs@eng-reg1.example.com>;tag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P.
To: <sip:701@eng-reg1.example.com>.
Contact: <sip:sip700_tbs@172.16.60.28:54936;ob>.
Call-ID: BUDvyYQczBQPRIOhj7t7FodP7675QPGi.
CSeq: 13862 INVITE.
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS.
Supported: replaces, 100rel, timer, norefersub.
Session-Expires: 1800.
Min-SE: 90.
User-Agent: CSipSimple_mecha-15/r2272.
Content-Type: application/sdp.
Content-Length:   340.
.
v=0.
o=- 3586685645 3586685645 IN IP4 172.16.60.28.
s=pjmedia.
c=IN IP4 172.16.60.28.
t=0 0.
m=audio 4004 RTP/AVP 99 0 8 101.
c=IN IP4 172.16.60.28.
a=rtcp:4005 IN IP4 172.16.60.28.
a=sendrecv.
a=rtpmap:99 SILK/24000.
a=fmtp:99 useinbandfec=0.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.


U 172.16.60.6:5060 -> 172.16.60.20:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 172.16.60.20;branch=z9hG4bKe51f.d2338795.0;received=172.16.60.20;rport=5060.
Via: SIP/2.0/UDP 172.16.60.28:54936;rport=54936;branch=z9hG4bKPjBsmwQX-oWu6srxf09JRx98W5IklAwE44.
Record-Route: <sip:172.16.60.20;lr=on;ftag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P;nat=yes>.
From: <sip:sip700_tbs@eng-reg1.example.com>;tag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P.
To: <sip:701@eng-reg1.example.com>.
Call-ID: BUDvyYQczBQPRIOhj7t7FodP7675QPGi.
CSeq: 13862 INVITE.
Server: Asterisk1.8.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:701@172.16.60.6:5060>.
Content-Length: 0.
.


U 172.16.60.6:5060 -> 172.16.60.20:5060
INVITE sip:sip701_tbs@172.16.60.20:5060 SIP/2.0.
Via: SIP/2.0/UDP 172.16.60.6:5060;branch=z9hG4bK0919ead7;rport.
Max-Forwards: 70.
From: "Alpha" <sip:700@172.16.60.6>;tag=as5e1a80d8.
To: <sip:sip701_tbs@172.16.60.20:5060>.
Contact: <sip:700@172.16.60.6:5060>.
Call-ID: 64a513d30fc6a51e54e8255b7169345c@172.16.60.6:5060.
CSeq: 102 INVITE.
User-Agent: Asterisk1.8.
Date: Wed, 28 Aug 2013 13:34:03 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 263.
.
v=0.
o=root 1276916964 1276916964 IN IP4 172.16.60.6.
s=Asterisk PBX 1.8.15-cert2.
c=IN IP4 172.16.60.6.
t=0 0.
m=audio 21930 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.


U 172.16.60.20:5060 -> 172.16.60.6:5060
SIP/2.0 100 trying -- your call is important to us.
Via: SIP/2.0/UDP 172.16.60.6:5060;branch=z9hG4bK0919ead7;rport=5060.
From: "Alpha" <sip:700@172.16.60.6>;tag=as5e1a80d8.
To: <sip:sip701_tbs@172.16.60.20:5060>.
Call-ID: 64a513d30fc6a51e54e8255b7169345c@172.16.60.6:5060.
CSeq: 102 INVITE.
Server: kamailio (4.0.3 (x86_64/linux)).
Content-Length: 0.
.


U 172.16.60.20:5060 -> 172.16.3.65:5060
INVITE sip:sip701_tbs@172.16.3.65:5060 SIP/2.0.
Record-Route: <sip:172.16.60.20;lr=on;ftag=as5e1a80d8;nat=yes>.
Via: SIP/2.0/UDP 172.16.60.20;branch=z9hG4bK9381.d7d662b.0.
Via: SIP/2.0/UDP 172.16.60.6:5060;branch=z9hG4bK0919ead7;rport=5060.
Max-Forwards: 16.
From: "Alpha" <sip:700@172.16.60.6>;tag=as5e1a80d8.
To: <sip:sip701_tbs@172.16.60.20:5060>.
Contact: <sip:700@172.16.60.6:5060>.
Call-ID: 64a513d30fc6a51e54e8255b7169345c@172.16.60.6:5060.
CSeq: 102 INVITE.
User-Agent: Asterisk1.8.
Date: Wed, 28 Aug 2013 13:34:03 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 263.
.
v=0.
o=root 1276916964 1276916964 IN IP4 172.16.60.6.
s=Asterisk PBX 1.8.15-cert2.
c=IN IP4 172.16.60.6.
t=0 0.
m=audio 21930 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.


U 172.16.3.65:5060 -> 172.16.60.20:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 172.16.60.20;received=172.16.60.20;branch=z9hG4bK9381.d7d662b.0.
Via: SIP/2.0/UDP 172.16.60.6:5060;rport=5060;branch=z9hG4bK0919ead7.
Record-Route: <sip:172.16.60.20;lr;ftag=as5e1a80d8;nat=yes>.
Call-ID: 64a513d30fc6a51e54e8255b7169345c@172.16.60.6:5060.
From: "Alpha" <sip:700@172.16.60.6>;tag=as5e1a80d8.
To: <sip:sip701_tbs@172.16.60.20>;tag=64c37e7e-19b8-46ae-ad0d-f2a4abe667bc.
CSeq: 102 INVITE.
Contact: <sip:sip701_tbs@172.16.60.20:5060>.
Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE.
Content-Length:  0.
.


U 172.16.60.20:5060 -> 172.16.60.6:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 172.16.60.6:5060;rport=5060;branch=z9hG4bK0919ead7.
Record-Route: <sip:172.16.60.20;lr;ftag=as5e1a80d8;nat=yes>.
Call-ID: 64a513d30fc6a51e54e8255b7169345c@172.16.60.6:5060.
From: "Alpha" <sip:700@172.16.60.6>;tag=as5e1a80d8.
To: <sip:sip701_tbs@172.16.60.20>;tag=64c37e7e-19b8-46ae-ad0d-f2a4abe667bc.
CSeq: 102 INVITE.
Contact: <sip:sip701_tbs@172.16.60.20:5060;alias=172.16.3.65~5060~1>.
Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE.
Content-Length:  0.
.


U 172.16.60.6:5060 -> 172.16.60.20:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 172.16.60.20;branch=z9hG4bKe51f.d2338795.0;received=172.16.60.20;rport=5060.
Via: SIP/2.0/UDP 172.16.60.28:54936;rport=54936;branch=z9hG4bKPjBsmwQX-oWu6srxf09JRx98W5IklAwE44.
Record-Route: <sip:172.16.60.20;lr=on;ftag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P;nat=yes>.
From: <sip:sip700_tbs@eng-reg1.example.com>;tag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P.
To: <sip:701@eng-reg1.example.com>;tag=as7e09c7c3.
Call-ID: BUDvyYQczBQPRIOhj7t7FodP7675QPGi.
CSeq: 13862 INVITE.
Server: Asterisk1.8.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:701@172.16.60.6:5060>.
Content-Length: 0.
.


U 172.16.60.20:5060 -> 172.16.60.28:54936
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 172.16.60.28:54936;rport=54936;branch=z9hG4bKPjBsmwQX-oWu6srxf09JRx98W5IklAwE44.
Record-Route: <sip:172.16.60.20;lr=on;ftag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P;nat=yes>.
From: <sip:sip700_tbs@eng-reg1.example.com>;tag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P.
To: <sip:701@eng-reg1.example.com>;tag=as7e09c7c3.
Call-ID: BUDvyYQczBQPRIOhj7t7FodP7675QPGi.
CSeq: 13862 INVITE.
Server: Asterisk1.8.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:701@172.16.60.6:5060>.
Content-Length: 0.
.


U 172.16.3.65:5060 -> 172.16.60.20:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 172.16.60.20;received=172.16.60.20;branch=z9hG4bK9381.d7d662b.0.
Via: SIP/2.0/UDP 172.16.60.6:5060;rport=5060;branch=z9hG4bK0919ead7.
Record-Route: <sip:172.16.60.20;lr;ftag=as5e1a80d8;nat=yes>.
Call-ID: 64a513d30fc6a51e54e8255b7169345c@172.16.60.6:5060.
From: "Alpha" <sip:700@172.16.60.6>;tag=as5e1a80d8.
To: <sip:sip701_tbs@172.16.60.20>;tag=64c37e7e-19b8-46ae-ad0d-f2a4abe667bc.
CSeq: 102 INVITE.
Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE.
Contact: <sip:sip701_tbs@172.16.60.20:5060>.
Supported: replaces, 100rel.
Content-Type: application/sdp.
Content-Length:   200.
.
v=0.
o=dresden 3586685646 1 IN IP4 172.16.3.65.
s=sflphone.
c=IN IP4 172.16.3.65.
t=0 0.
m=audio 41394 RTP/AVP 0.
a=rtpmap:0 PCMU/8000.
a=sendrecv.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.


U 172.16.60.20:5060 -> 172.16.60.6:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 172.16.60.6:5060;rport=5060;branch=z9hG4bK0919ead7.
Record-Route: <sip:172.16.60.20;lr;ftag=as5e1a80d8;nat=yes>.
Call-ID: 64a513d30fc6a51e54e8255b7169345c@172.16.60.6:5060.
From: "Alpha" <sip:700@172.16.60.6>;tag=as5e1a80d8.
To: <sip:sip701_tbs@172.16.60.20>;tag=64c37e7e-19b8-46ae-ad0d-f2a4abe667bc.
CSeq: 102 INVITE.
Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE.
Contact: <sip:sip701_tbs@172.16.60.20:5060;alias=172.16.3.65~5060~1>.
Supported: replaces, 100rel.
Content-Type: application/sdp.
Content-Length:   200.
.
v=0.
o=dresden 3586685646 1 IN IP4 172.16.3.65.
s=sflphone.
c=IN IP4 172.16.3.65.
t=0 0.
m=audio 41394 RTP/AVP 0.
a=rtpmap:0 PCMU/8000.
a=sendrecv.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.


=====
This ACK is never being sent back to the endpoint that answered the call (172.16.3.65):
=====

U 172.16.60.6:5060 -> 172.16.60.20:5060
ACK sip:sip701_tbs@172.16.60.20:5060;alias=172.16.3.65~5060~1 SIP/2.0.
Via: SIP/2.0/UDP 172.16.60.6:5060;branch=z9hG4bK70890881;rport.
Route: <sip:172.16.60.20;lr;ftag=as5e1a80d8;nat=yes>.
Max-Forwards: 70.
From: "Alpha" <sip:700@172.16.60.6>;tag=as5e1a80d8.
To: <sip:sip701_tbs@172.16.60.20:5060>;tag=64c37e7e-19b8-46ae-ad0d-f2a4abe667bc.
Contact: <sip:700@172.16.60.6:5060>.
Call-ID: 64a513d30fc6a51e54e8255b7169345c@172.16.60.6:5060.
CSeq: 102 ACK.
User-Agent: Asterisk1.8.
Content-Length: 0.
.


U 172.16.60.6:5060 -> 172.16.60.20:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 172.16.60.20;branch=z9hG4bKe51f.d2338795.0;received=172.16.60.20;rport=5060.
Via: SIP/2.0/UDP 172.16.60.28:54936;rport=54936;branch=z9hG4bKPjBsmwQX-oWu6srxf09JRx98W5IklAwE44.
Record-Route: <sip:172.16.60.20;lr=on;ftag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P;nat=yes>.
From: <sip:sip700_tbs@eng-reg1.example.com>;tag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P.
To: <sip:701@eng-reg1.example.com>;tag=as7e09c7c3.
Call-ID: BUDvyYQczBQPRIOhj7t7FodP7675QPGi.
CSeq: 13862 INVITE.
Server: Asterisk1.8.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:701@172.16.60.6:5060>.
Content-Type: application/sdp.
Content-Length: 261.
.
v=0.
o=root 115762304 115762304 IN IP4 172.16.60.6.
s=Asterisk PBX 1.8.15-cert2.
c=IN IP4 172.16.60.6.
t=0 0.
m=audio 24198 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.


U 172.16.60.20:5060 -> 172.16.60.28:54936
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 172.16.60.28:54936;rport=54936;branch=z9hG4bKPjBsmwQX-oWu6srxf09JRx98W5IklAwE44.
Record-Route: <sip:172.16.60.20;lr=on;ftag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P;nat=yes>.
From: <sip:sip700_tbs@eng-reg1.example.com>;tag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P.
To: <sip:701@eng-reg1.example.com>;tag=as7e09c7c3.
Call-ID: BUDvyYQczBQPRIOhj7t7FodP7675QPGi.
CSeq: 13862 INVITE.
Server: Asterisk1.8.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:701@172.16.60.6:5060>.
Content-Type: application/sdp.
Content-Length: 261.
.
v=0.
o=root 115762304 115762304 IN IP4 172.16.60.6.
s=Asterisk PBX 1.8.15-cert2.
c=IN IP4 172.16.60.6.
t=0 0.
m=audio 24198 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.


=====
This is the ACK that shows up on the loopback interface (I believe it should be going to 172.16.3.65):
=====

U 172.16.60.20:5060 -> 172.16.60.20:5060
ACK sip:172.16.60.20;lr;ftag=as5e1a80d8;nat=yes SIP/2.0.
Via: SIP/2.0/UDP 172.16.60.20;branch=z9hG4bKcydzigwkX.
Via: SIP/2.0/UDP 172.16.60.6:5060;branch=z9hG4bK70890881;rport=5060.
Max-Forwards: 16.
From: "Alpha" <sip:700@172.16.60.6>;tag=as5e1a80d8.
To: <sip:sip701_tbs@172.16.60.20:5060>;tag=64c37e7e-19b8-46ae-ad0d-f2a4abe667bc.
Contact: <sip:700@172.16.60.6:5060>.
Call-ID: 64a513d30fc6a51e54e8255b7169345c@172.16.60.6:5060.
CSeq: 102 ACK.
User-Agent: Asterisk1.8.
Content-Length: 0.
.


U 172.16.60.28:54936 -> 172.16.60.20:5060
ACK sip:701@172.16.60.6:5060 SIP/2.0.
Via: SIP/2.0/UDP 172.16.60.28:54936;rport;branch=z9hG4bKPjK4X6qtfWQUXsbgPP8D6j-bxR0KBd7r-x.
Max-Forwards: 70.
From: <sip:sip700_tbs@eng-reg1.example.com>;tag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P.
To: <sip:701@eng-reg1.example.com>;tag=as7e09c7c3.
Call-ID: BUDvyYQczBQPRIOhj7t7FodP7675QPGi.
CSeq: 13862 ACK.
Route: <sip:172.16.60.20;lr;ftag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P;nat=yes>.
Content-Length:  0.
.


U 172.16.60.20:5060 -> 172.16.60.6:5060
ACK sip:701@172.16.60.6:5060 SIP/2.0.
Via: SIP/2.0/UDP 172.16.60.20;branch=z9hG4bKcydzigwkX.
Via: SIP/2.0/UDP 172.16.60.28:54936;rport=54936;branch=z9hG4bKPjK4X6qtfWQUXsbgPP8D6j-bxR0KBd7r-x.
Max-Forwards: 16.
From: <sip:sip700_tbs@eng-reg1.example.com>;tag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P.
To: <sip:701@eng-reg1.example.com>;tag=as7e09c7c3.
Call-ID: BUDvyYQczBQPRIOhj7t7FodP7675QPGi.
CSeq: 13862 ACK.
Content-Length:  0.
.