[OpenSIPS-Users] sip message enters on bucle
Bogdan-Andrei Iancu
bogdan at opensips.org
Wed Apr 4 13:56:16 CEST 2012
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.
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20120404/05c2fcc9/attachment.htm>
More information about the Users
mailing list