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 [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
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