Jorge,
the message is not looping, it is retransmitting - it is something
different. OpenSIPS tries to open a new TCP conn to the destination (as
there is no existing one), but it fails in timeout as you cannot open a
TCP conn somewhere behind a NAT.
Regards,
Bogdan
On 04/04/2012 06:06 PM, Jorge Ortea wrote:
Hi Bogdan,
Is correct, Z.Z.Z.Z:5062 is a public adress behind a NAT. I have found
that opensips haven't this tcp connection, now this account has
changed the public adress.
But the sip messages keeps in the loop. It's like if Opensips is
looking for a tcp connection that it hasn't.... ?¿
Thanks.
Regards.
------------------------------------------------------------------------
Date: Wed, 4 Apr 2012 17:38:31 +0300
From: [email protected]
To: [email protected]
CC: [email protected]
Subject: Re: [OpenSIPS-Users] sip message enters on bucle
Hi Jorge,
So opensips tries to send the BYE to Z.Z.Z.Z:5062 via TCP (guess based
on Route hdrs), but nobody is listening on TCP - is this address
pointing behind a NAT ? why is not accepting a new TCP connection.
On the other side, what you can do is to reduce the timeout on TCP
connection, so opensips will react sooner:
http://www.opensips.org/Resources/DocsCoreFcn18#toc78
Regards,
Bogdan
On 04/04/2012 05:16 PM, Jorge Ortea wrote:
Hi Bogdan,
Exactly, is ready, OpenSIPS try to reach to destination but now
the account 2105 haven't the location: Z.Z.Z.Z:5062
In fact, when OpenSIPS try to reach to there, it write in log:
(this account uses TLS signaling)
Apr 4 10:14:17 alpha02 /usr/local/sbin/opensips[29503]: ::::::
BYE - from 911111111 to O2105 - Callid:
[email protected]
<mailto:[email protected]> - Source:
X.X.X.152
Apr 4 10:14:18 alpha02 /usr/local/sbin/opensips[29525]: ::::::
BYE - from 911111111 to O2105 - Callid:
[email protected]
<mailto:[email protected]> - Source:
X.X.X.152
Apr 4 10:14:19 alpha02 /usr/local/sbin/opensips[29497]: ::::::
BYE - from 911111111 to O2105 - Callid:
[email protected]
<mailto:[email protected]> - Source:
X.X.X.152
Apr 4 10:14:21 alpha02 /usr/local/sbin/opensips[29487]: ::::::
BYE - from 911111111 to O2105 - Callid:
[email protected]
<mailto:[email protected]> - Source:
X.X.X.152
Apr 4 10:14:25 alpha02 /usr/local/sbin/opensips[29511]: ::::::
BYE - from 911111111 to O2105 - Callid:
[email protected]
<mailto:[email protected]> - Source:
X.X.X.152
Apr 4 10:14:27 alpha02 /usr/local/sbin/opensips[29503]:
ERROR:core:tcp_blocking_connect: timeout 10 s elapsed from 10 s
Apr 4 10:14:27 alpha02 /usr/local/sbin/opensips[29503]:
ERROR:core:tcpconn_connect: tcp_blocking_connect failed
Apr 4 10:14:27 alpha02 /usr/local/sbin/opensips[29503]:
ERROR:core:tcp_send: connect failed
Apr 4 10:14:27 alpha02 /usr/local/sbin/opensips[29503]:
ERROR:tm:msg_send: tcp_send failed
Apr 4 10:14:27 alpha02 /usr/local/sbin/opensips[29503]:
ERROR:tm:t_forward_nonack: sending request failed
Thus, how can i detect and avoid this ??
Thanks.
Regards.
------------------------------------------------------------------------
Date: Wed, 4 Apr 2012 14:56:16 +0300
From: [email protected] <mailto:[email protected]>
To: [email protected] <mailto:[email protected]>
CC: [email protected] <mailto:[email protected]>
Subject: Re: [OpenSIPS-Users] sip message enters on bucle
Hi Jorge,
It looks like Asterisk generates the BYEs and retransmits it
because there is no reply coming back from opensips. Normally the
BYE is end 2 end replied (so the other end device should generate
the reply for BYE).
But looking at the 477 reply you get from OpenSIPS, I suspect that
OpenSIPS was trying to forward the BYE request (maybe via TCP),
got blocked and failed at the end - this failure resulted in the
477 reply.
Check the opensips logs to see error when processing the BYE.
Regards,
Bogdan
On 04/04/2012 11:42 AM, Jorge Ortea wrote:
Hi,
I have the follow VoIP platform; OpenSIPS 1.6.4.2-tls +
Mediaproxy 2.0 + a pair of Asterisks 1.4 (behind SER)
It works fine but sometimes a sip message enters on a loop.
Asterisk sends 5 sip messages at every turn
My logs in OpenSIPS:
Apr 4 10:14:17 alpha02 /usr/local/sbin/opensips[29503]:
:::::: BYE - from 911111111 to O2105 - Callid:
[email protected]
<mailto:[email protected]> - Source:
X.X.X.152
Apr 4 10:14:18 alpha02 /usr/local/sbin/opensips[29525]:
:::::: BYE - from 911111111 to O2105 - Callid:
[email protected]
<mailto:[email protected]> - Source:
X.X.X.152
Apr 4 10:14:19 alpha02 /usr/local/sbin/opensips[29497]:
:::::: BYE - from 911111111 to O2105 - Callid:
[email protected]
<mailto:[email protected]> - Source:
X.X.X.152
Apr 4 10:14:21 alpha02 /usr/local/sbin/opensips[29487]:
:::::: BYE - from 911111111 to O2105 - Callid:
[email protected]
<mailto:[email protected]> - Source:
X.X.X.152
Apr 4 10:14:25 alpha02 /usr/local/sbin/opensips[29511]:
:::::: BYE - from 911111111 to O2105 - Callid:
[email protected]
<mailto:[email protected]> - Source:
X.X.X.152
Sip messages in Asterisk *CLI> 'sip debug':
set_destination: Parsing
<sip:X.X.X.150;r2=on;lr=on;did=e25.7093e044> for address/port
to send to
set_destination: set destination to X.X.X.150, port 5060
Reliably Transmitting (no NAT) to X.X.X.150:5060:
BYE sip:[email protected]:5062;transport=tls SIP/2.0
Via: SIP/2.0/UDP X.X.X.152:5060;branch=z9hG4bK59044fff;rport
Route:
<sip:X.X.X.150;r2=on;lr=on;did=e25.7093e044>,<sip:Y.Y.Y.150:5061;transport=tls;r2=on;lr=on;did=e25.7093e044>
From: "911111111" <sip:[email protected]>;tag=as167eb28e
To: <sip:[email protected]>;tag=bcd482cd12b8a21i0
Call-ID: [email protected]
<mailto:[email protected]>
CSeq: 2874 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
Scheduling destruction of SIP dialog
'[email protected]
<mailto:[email protected]>' in 32000
ms (Method: REFER)
Retransmitting #1 (no NAT) to X.X.X.150:5060:
BYE sip:[email protected]:5062;transport=tls SIP/2.0
Via: SIP/2.0/UDP X.X.X.152:5060;branch=z9hG4bK59044fff;rport
Route:
<sip:X.X.X.150;r2=on;lr=on;did=e25.7093e044>,<sip:Y.Y.Y.150:5061;transport=tls;r2=on;lr=on;did=e25.7093e044>
From: "911111111" <sip:[email protected]>;tag=as167eb28e
To: <sip:[email protected]>;tag=bcd482cd12b8a21i0
Call-ID: [email protected]
<mailto:[email protected]>
CSeq: 2874 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
Retransmitting #2 (no NAT) to X.X.X.150:5060:
BYE sip:[email protected]:5062;transport=tls SIP/2.0
Via: SIP/2.0/UDP X.X.X.152:5060;branch=z9hG4bK59044fff;rport
Route:
<sip:X.X.X.150;r2=on;lr=on;did=e25.7093e044>,<sip:Y.Y.Y.150:5061;transport=tls;r2=on;lr=on;did=e25.7093e044>
From: "911111111" <sip:[email protected]>;tag=as167eb28e
To: <sip:[email protected]>;tag=bcd482cd12b8a21i0
Call-ID: [email protected]
<mailto:[email protected]>
CSeq: 2874 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
Retransmitting #3 (no NAT) to X.X.X.150:5060:
BYE sip:[email protected]:5062;transport=tls SIP/2.0
Via: SIP/2.0/UDP X.X.X.152:5060;branch=z9hG4bK59044fff;rport
Route:
<sip:X.X.X.150;r2=on;lr=on;did=e25.7093e044>,<sip:Y.Y.Y.150:5061;transport=tls;r2=on;lr=on;did=e25.7093e044>
From: "911111111" <sip:[email protected]>;tag=as167eb28e
To: <sip:[email protected]>;tag=bcd482cd12b8a21i0
Call-ID: [email protected]
<mailto:[email protected]>
CSeq: 2874 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
Retransmitting #4 (no NAT) to X.X.X.150:5060:
BYE sip:[email protected]:5062;transport=tls SIP/2.0
Via: SIP/2.0/UDP X.X.X.152:5060;branch=z9hG4bK59044fff;rport
Route:
<sip:X.X.X.150;r2=on;lr=on;did=e25.7093e044>,<sip:Y.Y.Y.150:5061;transport=tls;r2=on;lr=on;did=e25.7093e044>
From: "911111111" <sip:[email protected]>;tag=as167eb28e
To: <sip:[email protected]>;tag=bcd482cd12b8a21i0
Call-ID: [email protected]
<mailto:[email protected]>
CSeq: 2874 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from X.X.X.150:5060 --->
SIP/2.0 477 Send failed (477/TM)
Via: SIP/2.0/UDP X.X.X.152:5060;branch=z9hG4bK59044fff;rport=5060
From: "911111111" <sip:[email protected]>;tag=as167eb28e
To: <sip:[email protected]>;tag=bcd482cd12b8a21i0
Call-ID: [email protected]
<mailto:[email protected]>
CSeq: 2874 BYE
Server: OpenSIPS (1.6.4-2-tls (i386/linux))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
-- Incoming call: Got SIP response 477 "Send failed
(477/TM)" back from X.X.X.150
At the end, i have restart the asterisk to solve it. How can I
avoid it ?
Thanks.
Regards.
_______________________________________________
Users mailing list
[email protected] <mailto:[email protected]>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
_______________________________________________
Users mailing list
[email protected]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
