Re: [SR-Users] Problem with ACK - sr-users

6 Jun 2013

      Hello Daniel/Stoyan
Thanks for your reply, here is a full sip trace from the first INVITE
message to the last acknowledge which is sent to PGW.
192.168.10.189 ==> 81.21.38.34
INVITE sip:94294294@81.21.38.34 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7ddbce3f;rport
Max-Forwards: 70
From: "22498045" sip:22498045@192.168.10.189;tag=as181922af
To: sip:94294294@81.21.38.34
Contact: sip:22498045@192.168.10.189:5060
Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.21.0)
Date: Thu, 06 Jun 2013 09:40:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 1692903116 1692903116 IN IP4 192.168.10.189
s=Asterisk PBX 1.8.21.0
c=IN IP4 192.168.10.189
t=0 0
m=audio 12584 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
81.21.38.34 ==> 192.168.10.189
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7ddbce3f;rport=5060
From: "22498045" sip:22498045@192.168.10.189;tag=as181922af
To: sip:94294294@81.21.38.34
Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060
CSeq: 102 INVITE
Server: kamailio (4.0.1 (x86_64/linux))
Content-Length: 0
81.21.38.34 ==> 81.21.38.5
INVITE sip:3005A94294294@81.21.38.5 SIP/2.0
Record-Route: sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641
Via: SIP/2.0/UDP
81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0
Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7ddbce3f;rport=5060
Max-Forwards: 16
From: "22498045" sip:22498045@192.168.10.189;tag=as181922af
To: sip:94294294@81.21.38.34
Contact: sip:22498045@192.168.10.189:5060
Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.21.0)
Date: Thu, 06 Jun 2013 09:40:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 277
P-hint: outbound
v=0
o=root 1692903116 1692903116 IN IP4 81.21.38.34
s=Asterisk PBX 1.8.21.0
c=IN IP4 81.21.38.34
t=0 0
m=audio 38796 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
a=nortpproxy:yes
81.21.38.5 ==> 81.21.38.34
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0
Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7ddbce3f;rport=5060
From: "22498045" sip:22498045@192.168.10.189;tag=as181922af
To: sip:94294294@81.21.38.34
Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060
CSeq: 102 INVITE
Content-Length: 0
81.21.38.5 ==> 81.21.38.34
INVITE sip:7000A94294294@81.21.38.34:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP
81.21.38.5:5060;branch=z9hG4bKterm-1105c-22498045-94294294-67117
Record-Route: sip:94294294@81.21.38.5;pgw-call=call-2aa6
Record-Route: sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641
Via: SIP/2.0/UDP
81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0
Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7ddbce3f;rport=5060
Max-Forwards: 15
From: "22498045" sip:22498045@192.168.10.189;tag=as181922af
To: sip:94294294@81.21.38.34
Contact: sip:22498045@192.168.10.189:5060
Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.21.0)
Date: Thu, 06 Jun 2013 09:40:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 277
P-hint: outbound
v=0
o=root 1692903116 1692903116 IN IP4 81.21.38.34
s=Asterisk PBX 1.8.21.0
c=IN IP4 81.21.38.34
t=0 0
m=audio 38796 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
a=nortpproxy:yes
81.21.38.34 ==> 81.21.38.5
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP
81.21.38.5:5060;branch=z9hG4bKterm-1105c-22498045-94294294-67117
Via: SIP/2.0/UDP
81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0
Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7ddbce3f;rport=5060
From: "22498045" sip:22498045@192.168.10.189;tag=as181922af
To: sip:94294294@81.21.38.34
Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060
CSeq: 102 INVITE
Server: kamailio (4.0.1 (x86_64/linux))
Content-Length: 0
81.21.38.34==>81.21.38.55
INVITE sip:94294294@81.21.38.55 SIP/2.0
Record-Route: sip:81.21.38.34;lr=on;ftag=as181922af
Via: SIP/2.0/UDP
81.21.38.34;branch=z9hG4bKc526.ef179a29b1792073be0e0618cf49ac25.0
Via: SIP/2.0/UDP
81.21.38.5:5060;branch=z9hG4bKterm-1105c-22498045-94294294-67117
Record-Route: sip:94294294@81.21.38.5;pgw-call=call-2aa6
Record-Route: sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641
Via: SIP/2.0/UDP
81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0
Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7ddbce3f;rport=5060
Max-Forwards: 14
From: "22498045" sip:22498045@192.168.10.189;tag=as181922af
To: sip:94294294@81.21.38.34
Contact: sip:22498045@192.168.10.189:5060
Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.21.0)
Date: Thu, 06 Jun 2013 09:40:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 277
P-hint: outbound
P-hint: outbound
v=0
o=root 1692903116 1692903116 IN IP4 81.21.38.34
s=Asterisk PBX 1.8.21.0
c=IN IP4 81.21.38.34
t=0 0
m=audio 38796 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
a=nortpproxy:yes
81.21.38.55 ==> 81.21.38.34
SIP/2.0 200 OK
From: "22498045"sip:22498045@192.168.10.189;tag=as181922af
To: <sip:94294294@81.21.38.34
...
;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096
Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP
81.21.38.34;branch=z9hG4bKc526.ef179a29b1792073be0e0618cf49ac25.0
Via: SIP/2.0/UDP 81.21.38.5:5060
;branch=z9hG4bKterm-1105c-22498045-94294294-67117
Via: SIP/2.0/UDP
81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0
Via: SIP/2.0/UDP 192.168.10.189:5060;rport=5060;branch=z9hG4bK7ddbce3f
Supported: replaces
Record-Route: sip:81.21.38.34;lr=on;ftag=as181922af
Record-Route: sip:94294294@81.21.38.5;pgw-call=call-2aa6
Record-Route: sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641
Contact: sip:94294294@81.21.38.55
Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY, UPDATE
Content-Type: application/sdp
Content-Length: 242
v=0
o=Dialogic_SIP_CCLIB 64680888 64680889 IN IP4 81.21.38.55
s=Dialogic_SIP_CCLIB
c=IN IP4 81.21.38.55
t=0 0
m=audio 49156 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=sendrecv
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
81.21.38.34 ==> 81.21.38.5
SIP/2.0 200 OK
From: "22498045"sip:22498045@192.168.10.189;tag=as181922af
To: <sip:94294294@81.21.38.34
...
;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096
Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 81.21.38.5:5060
;branch=z9hG4bKterm-1105c-22498045-94294294-67117
Via: SIP/2.0/UDP
81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0
Via: SIP/2.0/UDP 192.168.10.189:5060;rport=5060;branch=z9hG4bK7ddbce3f
Supported: replaces
Record-Route: sip:81.21.38.34;lr=on;ftag=as181922af
Record-Route: sip:94294294@81.21.38.5;pgw-call=call-2aa6
Record-Route: sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641
Contact: sip:94294294@81.21.38.55
Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY, UPDATE
Content-Type: application/sdp
Content-Length: 260
v=0
o=Dialogic_SIP_CCLIB 64680888 64680889 IN IP4 81.21.38.34
s=Dialogic_SIP_CCLIB
c=IN IP4 81.21.38.34
t=0 0
m=audio 47332 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=sendrecv
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=nortpproxy:yes
81.21.38.5 ==> 81.21.38.34
SIP/2.0 200 OK
From: "22498045"sip:22498045@192.168.10.189;tag=as181922af
To: <sip:94294294@81.21.38.34
...
;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096
Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP
81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0
Via: SIP/2.0/UDP 192.168.10.189:5060;rport=5060;branch=z9hG4bK7ddbce3f
Supported: replaces
Record-Route: sip:81.21.38.34;lr=on;ftag=as181922af
Record-Route: sip:94294294@81.21.38.5;pgw-call=call-2aa6
Record-Route: sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641
Contact: sip:94294294@81.21.38.55
Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY, UPDATE
Content-Type: application/sdp
Content-Length: 260
v=0
o=Dialogic_SIP_CCLIB 64680888 64680889 IN IP4 81.21.38.34
s=Dialogic_SIP_CCLIB
c=IN IP4 81.21.38.34
t=0 0
m=audio 47332 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=sendrecv
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=nortpproxy:yes
81.21.38.34 ==> 192.168.10.189
SIP/2.0 200 OK
From: "22498045"sip:22498045@192.168.10.189;tag=as181922af
To: <sip:94294294@81.21.38.34
...
;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096
Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.10.189:5060;rport=5060;branch=z9hG4bK7ddbce3f
Supported: replaces
Record-Route: sip:81.21.38.34;lr=on;ftag=as181922af
Record-Route: sip:94294294@81.21.38.5;pgw-call=call-2aa6
Record-Route: sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641
Contact: sip:94294294@81.21.38.55
Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY, UPDATE
Content-Type: application/sdp
Content-Length: 260
v=0
o=Dialogic_SIP_CCLIB 64680888 64680889 IN IP4 81.21.38.34
s=Dialogic_SIP_CCLIB
c=IN IP4 81.21.38.34
t=0 0
m=audio 47332 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=sendrecv
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=nortpproxy:yes
#
U 192.168.10.189:5060 -> 81.21.38.34:5060
ACK sip:94294294@81.21.38.55 SIP/2.0*
Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK76f8f103;rport*
Route: sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641,<
sip:94294294@81.21.38.5
;pgw-call=call-2aa6>,sip:81.21.38.34;lr=on;ftag=as181922af*
Max-Forwards: 70*
From: "22498045" sip:22498045@192.168.10.189;tag=as181922af*
To: <sip:94294294@81.21.38.34
...
;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096*
Contact: sip:22498045@192.168.10.189:5060*
Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060*
CSeq: 102 ACK*
User-Agent: FPBX-2.8.1(1.8.21.0)*
Content-Length: 0*
81.21.38.34:5060 -> 81.21.38.5:5060
ACK sip:94294294@81.21.38.55 SIP/2.0*
Via: SIP/2.0/UDP
81.21.38.34;branch=z9hG4bKc526.9402c2edbc3ef96d9e405408364506a9.0*
Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK76f8f103;rport=5060*
Route: sip:94294294@81.21.38.5
;pgw-call=call-2aa6,sip:81.21.38.34;lr=on;ftag=as181922af*
Max-Forwards: 16*
From: "22498045" sip:22498045@192.168.10.189;tag=as181922af*
To: <sip:94294294@81.21.38.34
...
;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096*
Contact: sip:22498045@192.168.10.189:5060*
Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060*
CSeq: 102 ACK*
User-Agent: FPBX-2.8.1(1.8.21.0)*
Content-Length: 0*
Message: 3
Date: Thu, 06 Jun 2013 16:51:42 +0200
From: Daniel-Constantin Mierla miconda@gmail.com
To: "Kamailio (SER) - Users Mailing List"
        sr-users@lists.sip-router.org
Subject: Re: [SR-Users] Problem with ACK
Message-ID: 51B0A1FE.6040406@gmail.com
Content-Type: text/plain; charset="iso-8859-1"; Format="flowed"
On 6/6/13 4:34 PM, Stoyan Mihaylov wrote:
...
We had some similar problems.
But what was the actual problem? At least in the two ACKs provided
below, loose routing handling with looks correct.
Is something that Asterisk doesn't like?
Cheers,
Daniel
...
Our configuration is:
SIP client 1 <-> Kamailio <-> Asterisk <->Kamailio<->SIP client 2
My solution was to check $td and $si and if they are same as Kamailio,
to forward call to Asterisk.
Because I planed to use more then 1 Asterisk, I keep in variable which
one to use.
On Thu, Jun 6, 2013 at 5:28 PM, Daniel-Constantin Mierla
<miconda@gmail.com mailto:miconda@gmail.com> wrote:
Hello,

the incoming ACK has the top Route with lr parameter, meaning is
loose routing. By that, the proxy removes the top route header,
preserves the R-URI and sends to the URI in the next Route header.

From what I can see in the Route stack, it seems a spiral back to
the proxy because ip 81.21.38.34 is two times there.

If you can't sort it out, send the full SIP trace taken on the
proxy from the initial INVITE to the ACK. Then we can see how
Record-Route headers are set and the signaling flow.

Cheers,
Daniel

On 6/6/13 3:30 PM, phillman25 wrote:

...
Dear list further to the above problem i observed the following:

ACK message coming from PABX1:

U +0.001877 192.168.10.189:5060 <http://192.168.10.189:5060> ->
81.21.38.34:5060 <http://81.21.38.34:5060>
ACK sip:94294294@81.21.38.55 <mailto:sip%3A94294294@81.21.38.55>
SIP/2.0*
Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK76f8f103;rport*
Route:
<sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641>,<

sip:94294294@81.21.38.5
...
...
<mailto:sip%3A94294294@81.21.38.5>;pgw-call=call-2aa6>,<

sip:81.21.38.34;lr=on;ftag=as181922af>*
...
...
Max-Forwards: 70*
From: "22498045" <sip:22498045@192.168.10.189
<mailto:sip%3A22498045@192.168.10.189>>;tag=as181922af*
To: <sip:94294294@81.21.38.34
<mailto:sip%3A94294294@81.21.38.34>>;tag=3d95248-37261551-

13c4-50022-1c3096-5cc2673d-1c3096*
...
...
Contact: <sip:22498045@192.168.10.189:5060
<http://sip:22498045@192.168.10.189:5060>>*
Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060*
<mailto:696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060*>
CSeq: 102 ACK*
User-Agent: FPBX-2.8.1(1.8.21.0)*
Content-Length: 0*

ACK message sent to PGW from Kamailio1

U +0.001254 81.21.38.34:5060 <http://81.21.38.34:5060> ->
81.21.38.5:5060 <http://81.21.38.5:5060>
ACK sip:94294294@81.21.38.55 <mailto:sip%3A94294294@81.21.38.55>
SIP/2.0*
Via: SIP/2.0/UDP
81.21.38.34;branch=z9hG4bKc526.9402c2edbc3ef96d9e405408364506a9.0*
Via: SIP/2.0/UDP
192.168.10.189:5060;branch=z9hG4bK76f8f103;rport=5060*
Route: <sip:94294294@81.21.38.5
<mailto:sip%3A94294294@81.21.38.5>;pgw-call=call-2aa6>,<

sip:81.21.38.34;lr=on;ftag=as181922af>*
...
...
Max-Forwards: 16*
From: "22498045" <sip:22498045@192.168.10.189
<mailto:sip%3A22498045@192.168.10.189>>;tag=as181922af*
To: <sip:94294294@81.21.38.34
<mailto:sip%3A94294294@81.21.38.34>>;tag=3d95248-37261551-

13c4-50022-1c3096-5cc2673d-1c3096*
...
...
Contact: <sip:22498045@192.168.10.189:5060
<http://sip:22498045@192.168.10.189:5060>>*
Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060*
<mailto:696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060*>
CSeq: 102 ACK*
User-Agent: FPBX-2.8.1(1.8.21.0)*
Content-Length: 0*

Shouldn't the ACK  message to the PGW have the header ACK
sip:94294294@81.21.38.5
<mailto:sip%3A94294294@81.21.38.5>;pgw-call=call-2aa6 and the
Route: <sip:81.21.38.34;lr=on;ftag=as181922af>*   ???

Your help is much appreciated!!

Phillip

On Thu, Jun 6, 2013 at 12:26 PM, phillman25 <phillman25@gmail.com
<mailto:phillman25@gmail.com>> wrote:

    Dear List

    I upgraded from Kamailio v 3.3 to 4.0.1 and am now facing an
    issue for the below scenario:

    PABX1 ==> Kamailio1 ==> Cisco PGW ==> Kamailio1 ==> PABX2

    I understand that this is a hairpin scenario but was working
    normally on v 3.3.

    Checking in the syslog i see:
    ERROR: <core> [receive.c:230]: ERROR: receive_msg: no via
    found in reply

    Checking the sip trace i see that when calling from PABX1 to
    PABX2. After PABX2 answers and the the 200 OK  is eventually
    sent  to PABX1 , PABX1 answers with ACK but seems like its
    not sent back to PABX2  as a result PABX resends a 200 OK and
    the cycle continues until PABX2 sends a BYE message. Please
    see below the ACK received from PABX1:

    ACK sip:94294294@81.21.38.55
    <mailto:sip%3A94294294@81.21.38.55> SIP/2.0
    Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK6bffe37c;rport
    Route:
    <sip:81.21.38.34;lr=on;ftag=as1cd4f8f1;did=e36.c471>,<

sip:94294294@81.21.38.5
...
...
    <mailto:sip%3A94294294@81.21.38.5>;pgw-call=call-26eb>,<

sip:81.21.38.34;lr=on;ftag=as1cd4f8f1>
...
...
    Max-Forwards: 70
    From: "22498045" <sip:22498045@192.168.10.189
    <mailto:sip%3A22498045@192.168.10.189>>;tag=as1cd4f8f1
    To: <sip:94294294@81.21.38.34
    <mailto:sip%3A94294294@81.21.38.34>>;tag=3d94f08-37261551-

13c4-50022-1c1e67-87fe958-1c1e67
...
...
    Contact: <sip:22498045@192.168.10.189:5060
    <http://sip:22498045@192.168.10.189:5060>>
    Call-ID: 03042a717e27a87e759f7f4879e70377@192.168.10.189:5060
    <http://03042a717e27a87e759f7f4879e70377@192.168.10.189:5060>
    CSeq: 102 ACK
    User-Agent: FPBX-2.8.1(1.8.21.0)
    Content-Length: 0

    Is there an issue with the above ACK message? Is there any
    way to solve this issue quickly perhaps by disabling loose route?
    I have observed that this issue occurs only when hairpinned.

    Thanking you in advance!

    Phillip