[OpenSIPS-Users] sip message enters on bucle

Jorge Ortea darham at hotmail.com
Wed Apr 4 16:16:04 CEST 2012



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 		 	   		  
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