[OpenSIPS-Users] sip message enters on bucle
Jorge Ortea
darham at hotmail.com
Wed Apr 4 17:06:17 CEST 2012
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: bogdan at opensips.org
To: darham at hotmail.com
CC: users at lists.opensips.org
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: 5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
- Source: X.X.X.152
Apr 4 10:14:18 alpha02 /usr/local/sbin/opensips[29525]: ::::::
BYE - from 911111111 to O2105 - Callid: 5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
- Source: X.X.X.152
Apr 4 10:14:19 alpha02 /usr/local/sbin/opensips[29497]: ::::::
BYE - from 911111111 to O2105 - Callid: 5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
- Source: X.X.X.152
Apr 4 10:14:21 alpha02 /usr/local/sbin/opensips[29487]: ::::::
BYE - from 911111111 to O2105 - Callid: 5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
- Source: X.X.X.152
Apr 4 10:14:25 alpha02 /usr/local/sbin/opensips[29511]: ::::::
BYE - from 911111111 to O2105 - Callid: 5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
- 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: bogdan at opensips.org
To: users at lists.opensips.org
CC: darham at hotmail.com
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: 5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
- Source: X.X.X.152
Apr 4 10:14:18 alpha02 /usr/local/sbin/opensips[29525]:
:::::: BYE - from 911111111 to O2105 - Callid: 5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
- Source: X.X.X.152
Apr 4 10:14:19 alpha02 /usr/local/sbin/opensips[29497]:
:::::: BYE - from 911111111 to O2105 - Callid: 5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
- Source: X.X.X.152
Apr 4 10:14:21 alpha02 /usr/local/sbin/opensips[29487]:
:::::: BYE - from 911111111 to O2105 - Callid: 5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
- Source: X.X.X.152
Apr 4 10:14:25 alpha02 /usr/local/sbin/opensips[29511]:
:::::: BYE - from 911111111 to O2105 - Callid: 5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
- 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:2105 at Z.Z.Z.Z: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:911111111 at X.X.X.152>;tag=as167eb28e
To: <sip:O2105 at X.X.X.150>;tag=bcd482cd12b8a21i0
Call-ID: 5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
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 '5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152'
in 32000 ms (Method: REFER)
Retransmitting #1 (no NAT) to X.X.X.150:5060:
BYE sip:2105 at Z.Z.Z.Z: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:911111111 at X.X.X.152>;tag=as167eb28e
To: <sip:O2105 at X.X.X.150>;tag=bcd482cd12b8a21i0
Call-ID: 5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
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:2105 at Z.Z.Z.Z: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:911111111 at X.X.X.152>;tag=as167eb28e
To: <sip:O2105 at X.X.X.150>;tag=bcd482cd12b8a21i0
Call-ID: 5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
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:2105 at Z.Z.Z.Z: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:911111111 at X.X.X.152>;tag=as167eb28e
To: <sip:O2105 at X.X.X.150>;tag=bcd482cd12b8a21i0
Call-ID: 5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
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:2105 at Z.Z.Z.Z: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:911111111 at X.X.X.152>;tag=as167eb28e
To: <sip:O2105 at X.X.X.150>;tag=bcd482cd12b8a21i0
Call-ID: 5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
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:911111111 at X.X.X.152>;tag=as167eb28e
To: <sip:O2105 at X.X.X.150>;tag=bcd482cd12b8a21i0
Call-ID: 5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
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.
_______________________________________________
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--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
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