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:
[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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--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
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