[OpenSIPS-Users] sip message enters on bucle

Bogdan-Andrei Iancu bogdan at opensips.org
Wed Apr 4 16:38:31 CEST 2012


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 
> <mailto: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 
> <mailto: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 
> <mailto: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 
> <mailto: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 
> <mailto: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
>     <mailto: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
>     <mailto: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
>     <mailto: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
>     <mailto: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
>     <mailto: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
>     <mailto: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
>     <mailto: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
>     <mailto: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
>     <mailto: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
>     <mailto: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
>     <mailto: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
>     <mailto: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.
>
>
>
>     _______________________________________________
>     Users mailing list
>     Users at lists.opensips.org  <mailto:Users at lists.opensips.org>
>     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

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