[SR-Users] Problem with ACK
phillman25
phillman25 at gmail.com
Thu Jun 6 17:37:33 CEST 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 at 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 at 192.168.10.189>;tag=as181922af
To: <sip:94294294 at 81.21.38.34>
Contact: <sip:22498045 at 192.168.10.189:5060>
Call-ID: 696cfa577f42395767bc812e7e8a38a5 at 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 at 192.168.10.189>;tag=as181922af
To: <sip:94294294 at 81.21.38.34>
Call-ID: 696cfa577f42395767bc812e7e8a38a5 at 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 at 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 at 192.168.10.189>;tag=as181922af
To: <sip:94294294 at 81.21.38.34>
Contact: <sip:22498045 at 192.168.10.189:5060>
Call-ID: 696cfa577f42395767bc812e7e8a38a5 at 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 at 192.168.10.189>;tag=as181922af
To: <sip:94294294 at 81.21.38.34>
Call-ID: 696cfa577f42395767bc812e7e8a38a5 at 192.168.10.189:5060
CSeq: 102 INVITE
Content-Length: 0
81.21.38.5 ==> 81.21.38.34
INVITE sip:7000A94294294 at 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 at 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 at 192.168.10.189>;tag=as181922af
To: <sip:94294294 at 81.21.38.34>
Contact: <sip:22498045 at 192.168.10.189:5060>
Call-ID: 696cfa577f42395767bc812e7e8a38a5 at 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 at 192.168.10.189>;tag=as181922af
To: <sip:94294294 at 81.21.38.34>
Call-ID: 696cfa577f42395767bc812e7e8a38a5 at 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 at 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 at 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 at 192.168.10.189>;tag=as181922af
To: <sip:94294294 at 81.21.38.34>
Contact: <sip:22498045 at 192.168.10.189:5060>
Call-ID: 696cfa577f42395767bc812e7e8a38a5 at 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 at 192.168.10.189>;tag=as181922af
To: <sip:94294294 at 81.21.38.34
>;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096
Call-ID: 696cfa577f42395767bc812e7e8a38a5 at 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 at 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 at 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 at 192.168.10.189>;tag=as181922af
To: <sip:94294294 at 81.21.38.34
>;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096
Call-ID: 696cfa577f42395767bc812e7e8a38a5 at 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 at 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 at 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 at 192.168.10.189>;tag=as181922af
To: <sip:94294294 at 81.21.38.34
>;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096
Call-ID: 696cfa577f42395767bc812e7e8a38a5 at 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 at 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 at 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 at 192.168.10.189>;tag=as181922af
To: <sip:94294294 at 81.21.38.34
>;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096
Call-ID: 696cfa577f42395767bc812e7e8a38a5 at 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 at 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 at 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 at 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 at 81.21.38.5
;pgw-call=call-2aa6>,<sip:81.21.38.34;lr=on;ftag=as181922af>*
Max-Forwards: 70*
From: "22498045" <sip:22498045 at 192.168.10.189>;tag=as181922af*
To: <sip:94294294 at 81.21.38.34
>;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096*
Contact: <sip:22498045 at 192.168.10.189:5060>*
Call-ID: 696cfa577f42395767bc812e7e8a38a5 at 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 at 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 at 81.21.38.5
;pgw-call=call-2aa6>,<sip:81.21.38.34;lr=on;ftag=as181922af>*
Max-Forwards: 16*
From: "22498045" <sip:22498045 at 192.168.10.189>;tag=as181922af*
To: <sip:94294294 at 81.21.38.34
>;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096*
Contact: <sip:22498045 at 192.168.10.189:5060>*
Call-ID: 696cfa577f42395767bc812e7e8a38a5 at 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 at gmail.com>
To: "Kamailio (SER) - Users Mailing List"
<sr-users at lists.sip-router.org>
Subject: Re: [SR-Users] Problem with ACK
Message-ID: <51B0A1FE.6040406 at 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 at gmail.com <mailto:miconda at 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 at 81.21.38.55 <mailto:sip%3A94294294 at 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 at 81.21.38.5
>> <mailto:sip%3A94294294 at 81.21.38.5>;pgw-call=call-2aa6>,<
sip:81.21.38.34;lr=on;ftag=as181922af>*
>> Max-Forwards: 70*
>> From: "22498045" <sip:22498045 at 192.168.10.189
>> <mailto:sip%3A22498045 at 192.168.10.189>>;tag=as181922af*
>> To: <sip:94294294 at 81.21.38.34
>> <mailto:sip%3A94294294 at 81.21.38.34>>;tag=3d95248-37261551-
13c4-50022-1c3096-5cc2673d-1c3096*
>> Contact: <sip:22498045 at 192.168.10.189:5060
>> <http://sip:22498045@192.168.10.189:5060>>*
>> Call-ID: 696cfa577f42395767bc812e7e8a38a5 at 192.168.10.189:5060*
>> <mailto:696cfa577f42395767bc812e7e8a38a5 at 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 at 81.21.38.55 <mailto:sip%3A94294294 at 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 at 81.21.38.5
>> <mailto:sip%3A94294294 at 81.21.38.5>;pgw-call=call-2aa6>,<
sip:81.21.38.34;lr=on;ftag=as181922af>*
>> Max-Forwards: 16*
>> From: "22498045" <sip:22498045 at 192.168.10.189
>> <mailto:sip%3A22498045 at 192.168.10.189>>;tag=as181922af*
>> To: <sip:94294294 at 81.21.38.34
>> <mailto:sip%3A94294294 at 81.21.38.34>>;tag=3d95248-37261551-
13c4-50022-1c3096-5cc2673d-1c3096*
>> Contact: <sip:22498045 at 192.168.10.189:5060
>> <http://sip:22498045@192.168.10.189:5060>>*
>> Call-ID: 696cfa577f42395767bc812e7e8a38a5 at 192.168.10.189:5060*
>> <mailto:696cfa577f42395767bc812e7e8a38a5 at 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 at 81.21.38.5
>> <mailto:sip%3A94294294 at 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 at gmail.com
>> <mailto:phillman25 at 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 at 81.21.38.55
>> <mailto:sip%3A94294294 at 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 at 81.21.38.5
>> <mailto:sip%3A94294294 at 81.21.38.5>;pgw-call=call-26eb>,<
sip:81.21.38.34;lr=on;ftag=as1cd4f8f1>
>> Max-Forwards: 70
>> From: "22498045" <sip:22498045 at 192.168.10.189
>> <mailto:sip%3A22498045 at 192.168.10.189>>;tag=as1cd4f8f1
>> To: <sip:94294294 at 81.21.38.34
>> <mailto:sip%3A94294294 at 81.21.38.34>>;tag=3d94f08-37261551-
13c4-50022-1c1e67-87fe958-1c1e67
>> Contact: <sip:22498045 at 192.168.10.189:5060
>> <http://sip:22498045@192.168.10.189:5060>>
>> Call-ID: 03042a717e27a87e759f7f4879e70377 at 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
>>
>>
>>
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