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
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@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
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@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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@gmail.comwrote:
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@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
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
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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 _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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@gmail.comwrote:
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> 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@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
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
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list 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 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
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-
Daniel-Constantin Mierla -
phillman25 -
Stoyan Mihaylov