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Hi,<br><br>I have the follow VoIP platform; OpenSIPS 1.6.4.2-tls + Mediaproxy 2.0 + a pair of Asterisks 1.4 (behind SER)<br><br>It works fine but sometimes a sip message enters on a loop. <span id="result_box" class="short_text" lang="en"><span class="hps">Asterisk</span> <span class="hps">sends</span> <span class="hps"></span></span><span id="result_box" class="short_text" lang="en"><span class="hps"></span><span class="hps">5 sip messages</span> <span class="hps">at every turn</span></span><br><br><br>My logs in OpenSIPS:<br><br>Apr 4 10:14:17 alpha02 /usr/local/sbin/opensips[29503]: :::::: BYE - from 911111111 to O2105 - Callid: 5b62cc795e6be4ea3fa9a26e543e3622@X.X.X.152 - Source: X.X.X.152<br>Apr 4 10:14:18 alpha02 /usr/local/sbin/opensips[29525]: :::::: BYE - from 911111111 to O2105 - Callid: 5b62cc795e6be4ea3fa9a26e543e3622@X.X.X.152 - Source: X.X.X.152<br>Apr 4 10:14:19 alpha02 /usr/local/sbin/opensips[29497]: :::::: BYE - from 911111111 to O2105 - Callid: 5b62cc795e6be4ea3fa9a26e543e3622@X.X.X.152 - Source: X.X.X.152<br>Apr 4 10:14:21 alpha02 /usr/local/sbin/opensips[29487]: :::::: BYE - from 911111111 to O2105 - Callid: 5b62cc795e6be4ea3fa9a26e543e3622@X.X.X.152 - Source: X.X.X.152<br>Apr 4 10:14:25 alpha02 /usr/local/sbin/opensips[29511]: :::::: BYE - from 911111111 to O2105 - Callid: 5b62cc795e6be4ea3fa9a26e543e3622@X.X.X.152 - Source: X.X.X.152<br> <br><br><br>Sip messages in Asterisk *CLI> 'sip debug':<br><br>set_destination: Parsing <sip:X.X.X.150;r2=on;lr=on;did=e25.7093e044> for address/port to send to<br>set_destination: set destination to X.X.X.150, port 5060<br>Reliably Transmitting (no NAT) to X.X.X.150:5060:<br>BYE sip:2105@Z.Z.Z.Z:5062;transport=tls SIP/2.0<br>Via: SIP/2.0/UDP X.X.X.152:5060;branch=z9hG4bK59044fff;rport<br>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><br>From: "911111111" <sip:911111111@X.X.X.152>;tag=as167eb28e<br>To: <sip:O2105@X.X.X.150>;tag=bcd482cd12b8a21i0<br>Call-ID: 5b62cc795e6be4ea3fa9a26e543e3622@X.X.X.152<br>CSeq: 2874 BYE<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70<br>X-Asterisk-HangupCause: Normal Clearing<br>X-Asterisk-HangupCauseCode: 16<br>Content-Length: 0<br><br><br>---<br>Scheduling destruction of SIP dialog '5b62cc795e6be4ea3fa9a26e543e3622@X.X.X.152' in 32000 ms (Method: REFER)<br>Retransmitting #1 (no NAT) to X.X.X.150:5060:<br>BYE sip:2105@Z.Z.Z.Z:5062;transport=tls SIP/2.0<br>Via: SIP/2.0/UDP X.X.X.152:5060;branch=z9hG4bK59044fff;rport<br>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><br>From: "911111111" <sip:911111111@X.X.X.152>;tag=as167eb28e<br>To: <sip:O2105@X.X.X.150>;tag=bcd482cd12b8a21i0<br>Call-ID: 5b62cc795e6be4ea3fa9a26e543e3622@X.X.X.152<br>CSeq: 2874 BYE<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70<br>X-Asterisk-HangupCause: Normal Clearing<br>X-Asterisk-HangupCauseCode: 16<br>Content-Length: 0<br><br><br>---<br>Retransmitting #2 (no NAT) to X.X.X.150:5060:<br>BYE sip:2105@Z.Z.Z.Z:5062;transport=tls SIP/2.0<br>Via: SIP/2.0/UDP X.X.X.152:5060;branch=z9hG4bK59044fff;rport<br>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><br>From: "911111111" <sip:911111111@X.X.X.152>;tag=as167eb28e<br>To: <sip:O2105@X.X.X.150>;tag=bcd482cd12b8a21i0<br>Call-ID: 5b62cc795e6be4ea3fa9a26e543e3622@X.X.X.152<br>CSeq: 2874 BYE<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70<br>X-Asterisk-HangupCause: Normal Clearing<br>X-Asterisk-HangupCauseCode: 16<br>Content-Length: 0<br><br><br>---<br>Retransmitting #3 (no NAT) to X.X.X.150:5060:<br>BYE sip:2105@Z.Z.Z.Z:5062;transport=tls SIP/2.0<br>Via: SIP/2.0/UDP X.X.X.152:5060;branch=z9hG4bK59044fff;rport<br>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><br>From: "911111111" <sip:911111111@X.X.X.152>;tag=as167eb28e<br>To: <sip:O2105@X.X.X.150>;tag=bcd482cd12b8a21i0<br>Call-ID: 5b62cc795e6be4ea3fa9a26e543e3622@X.X.X.152<br>CSeq: 2874 BYE<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70<br>X-Asterisk-HangupCause: Normal Clearing<br>X-Asterisk-HangupCauseCode: 16<br>Content-Length: 0<br><br><br>---<br>Retransmitting #4 (no NAT) to X.X.X.150:5060:<br>BYE sip:2105@Z.Z.Z.Z:5062;transport=tls SIP/2.0<br>Via: SIP/2.0/UDP X.X.X.152:5060;branch=z9hG4bK59044fff;rport<br>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><br>From: "911111111" <sip:911111111@X.X.X.152>;tag=as167eb28e<br>To: <sip:O2105@X.X.X.150>;tag=bcd482cd12b8a21i0<br>Call-ID: 5b62cc795e6be4ea3fa9a26e543e3622@X.X.X.152<br>CSeq: 2874 BYE<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70<br>X-Asterisk-HangupCause: Normal Clearing<br>X-Asterisk-HangupCauseCode: 16<br>Content-Length: 0<br><br><br>---<br><br><--- SIP read from X.X.X.150:5060 ---><br>SIP/2.0 477 Send failed (477/TM)<br>Via: SIP/2.0/UDP X.X.X.152:5060;branch=z9hG4bK59044fff;rport=5060<br>From: "911111111" <sip:911111111@X.X.X.152>;tag=as167eb28e<br>To: <sip:O2105@X.X.X.150>;tag=bcd482cd12b8a21i0<br>Call-ID: 5b62cc795e6be4ea3fa9a26e543e3622@X.X.X.152<br>CSeq: 2874 BYE<br>Server: OpenSIPS (1.6.4-2-tls (i386/linux))<br>Content-Length: 0<br><br><br><-------------><br>--- (8 headers 0 lines) ---<br>SIP Response message for INCOMING dialog BYE arrived<br> -- Incoming call: Got SIP response 477 "Send failed (477/TM)" back from X.X.X.150<br><br><br><br>At the end, i have restart the asterisk to solve it. How can I avoid it ?<br><br><br>Thanks.<br>Regards.<br><br><br> </div></body>
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