Hi Bogdan,
Ok, now we known that is happening. But, is it logic? or is it a bug in 1.6.4.2
version?
Curiously this does not happen with UDP signaling.
Thanks.
Regards.
Date: Wed, 4 Apr 2012 18:19:04 +0300
From: [email protected]
To: [email protected]
CC: [email protected]
Subject: Re: [OpenSIPS-Users] sip message enters on bucle
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]
- Source: X.X.X.152
Apr 4 10:14:18 alpha02 /usr/local/sbin/opensips[29525]:
:::::: BYE - from 911111111 to O2105 - Callid:
[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]
- Source: X.X.X.152
Apr 4 10:14:21 alpha02 /usr/local/sbin/opensips[29487]:
:::::: BYE - from 911111111 to O2105 - Callid:
[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]
- 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]
To: [email protected]
CC: [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]
- Source: X.X.X.152
Apr 4 10:14:18 alpha02
/usr/local/sbin/opensips[29525]: :::::: BYE - from
911111111 to O2105 - Callid:
[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]
- Source: X.X.X.152
Apr 4 10:14:21 alpha02
/usr/local/sbin/opensips[29487]: :::::: BYE - from
911111111 to O2105 - Callid:
[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]
- 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]
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]'
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]
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]
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]
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]
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]
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
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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
