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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