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
<mico...@gmail.com <mailto:mico...@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 <phillma...@gmail.com
<mailto:phillma...@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
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--
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 *
_______________________________________________
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list
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_______________________________________________
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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://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *
_______________________________________________
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