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
>>
>>
>>