[OpenSIPS-Users] sip message enters on bucle

[Jorge Ortea]

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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