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.
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--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com _______________________________________________
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