6 Jun
2013
6 Jun
'13
5:26 a.m.
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 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
;pgw-call=call-26eb>,<sip:81.21.38.34;lr=on;ftag=as1cd4f8f1>
Max-Forwards: 70
From: "22498045" <sip:22498045@192.168.10.189>;tag=as1cd4f8f1
To: <sip:94294294@81.21.38.34
;tag=3d94f08-37261551-13c4-50022-1c1e67-87fe958-1c1e67
Contact: <sip:22498045@192.168.10.189:5060>
Call-ID: 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
6 Jun
6 Jun
9:30 a.m.
Dear list further to the above problem i observed the following:
ACK message coming from PABX1:
U +0.001877 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*
ACK message sent to PGW from Kamailio1
U +0.001254 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*
Shouldn't the ACK message to the PGW have the header ACK
sip:94294294@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(a)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 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
;pgw-call=call-26eb>,<sip:81.21.38.34;lr=on;ftag=as1cd4f8f1>
Max-Forwards: 70
From: "22498045" <sip:22498045@192.168.10.189>;tag=as1cd4f8f1
To: <sip:94294294@81.21.38.34
;tag=3d94f08-37261551-13c4-50022-1c1e67-87fe958-1c1e67
Contact:
<sip:22498045@192.168.10.189:5060>
Call-ID: 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
10:28 a.m.
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*
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*
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(a)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
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *
10:34 a.m.
We had some similar problems. 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(a)gmail.com>wrote;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 -> 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*
ACK message sent to PGW from Kamailio1
U +0.001254 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*
Shouldn't the ACK message to the PGW have the header ACK
sip:94294294@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(a)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 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;pgw-call=call-26eb>,
<sip:81.21.38.34;lr=on;ftag=as1cd4f8f1>
Max-Forwards: 70
From: "22498045" <sip:22498045@192.168.10.189>;tag=as1cd4f8f1
To: <sip:94294294@81.21.38.34
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
;tag=3d94f08-37261551-13c4-50022-1c1e67-87fe958-1c1e67
Contact:
<sip:22498045@192.168.10.189:5060>
Call-ID: 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
10:51 a.m.
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(a)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:
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *
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(a)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
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla -http://www.asipto.com
http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda>
-http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
*http://asipto.com/u/katu *
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
11:06 a.m.
I dont know what caused problem. I just found working solution. I used
fireshark to get messages, and I saw that some ACK and BYE messages
"reenter" kamailio and keep growing - as I remember (not sure). As I
remember (not sure) - VIA started to grow for next messages. For me it
looked like message for some reason reenter kamailio, adding new VIA record.
My whole solution is:
If I receive ACK or BYE message, I process them next way:
route[ACKBYE] {
#!ifdef WITH_MYFORWARD
if(($sht(forw=>$ft))=~$td){
$du=$sht(forw=>$ft);
}else if((($td=="sip.OurCompany.com
")||($td=="xxx.xxx.xxx.xxx"))&&($si=="xxx.xxx.xxx.xxx")){
$du=$sht(forw=>$ft);
return;
}
#!endif
return;
}
Of course: sip.OurCompany.com=xxx.xxx.xxx.xxx
Here I initialize $sht(forw=>$ft)
route[PSTNINVITE] {
#!ifdef WITH_MYFORWARD
if(is_method("INVITE")){
ds_select_dst("1","4");
$sht(forw=>$ft)=$du;
sl_send_reply("100","Trying");
route(RELAY);
exit();
}
#!endif
return;
}
On Thu, Jun 6, 2013 at 5:51 PM, Daniel-Constantin Mierla
<miconda(a)gmail.com>wrote;wrote:
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(a)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 -> 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
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
;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*
ACK message sent to PGW from Kamailio1
U +0.001254 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*
Shouldn't the ACK message to the PGW have the header ACK
sip:94294294@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(a)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 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;pgw-call=call-26eb>,
<sip:81.21.38.34;lr=on;ftag=as1cd4f8f1>
Max-Forwards: 70
From: "22498045" <sip:22498045@192.168.10.189>;tag=as1cd4f8f1
To: <sip:94294294@81.21.38.34
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
;tag=3d94f08-37261551-13c4-50022-1c1e67-87fe958-1c1e67
Contact:
<sip:22498045@192.168.10.189:5060>
Call-ID: 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
4273
days inactive
4273
days old
5 comments
3 participants
participants (3)
-
Daniel-Constantin Mierla -
phillman25 -
Stoyan Mihaylov