[asterisk-bugs] [Asterisk 0015890]: 1.6.1.5 - "Ghost" channels
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri Sep 18 13:02:49 CDT 2009
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=15890
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Reported By: simonoch
Assigned To:
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Project: Asterisk
Issue ID: 15890
Category: Channels/General
Reproducibility: always
Severity: minor
Priority: normal
Status: feedback
Asterisk Version: 1.6.1.5
JIRA:
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2009-09-14 07:41 CDT
Last Modified: 2009-09-18 13:02 CDT
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Summary: 1.6.1.5 - "Ghost" channels
Description:
We seems to have ghost channels. We don't understand what's the problem.
# (extract) sip show channels
Peer User/ANR Call ID Format Hold
Last Message Expiry
82.241.145.220 103 6cb5e7541370ff7 0x0 (nothing) No
Init: INVITE
82.241.145.220 103 5f8a19a14b48234 0x0 (nothing) No
Init: INVITE
82.241.145.220 103 659369d21a838aa 0x0 (nothing) No
Init: INVITE
82.241.145.220 103 434b1ec828bc2e9 0x100 (g729) No
Tx: ACK
82.241.145.220 103 301947390a5111f 0x0 (nothing) No
Init: INVITE
82.241.145.220 103 524b5b7f6289b7a 0x0 (nothing) No
Init: INVITE
82.241.145.220 103 535d242452bdd48 0x0 (nothing) No
Init: INVITE
# sip show history 659369d21a838aa
* SIP Call
1. ReliableXmit timeout
2. ReliableXmit timeout
3. ReliableXmit timeout
4. ReliableXmit timeout
5. ReliableXmit timeout
6. ReliableXmit timeout
7. ReliableXmit timeout
8. ReliableXmit timeout
9. ReliableXmit timeout
10. ReliableXmit timeout
11. ReliableXmit timeout
12. ReliableXmit timeout
13. ReliableXmit timeout
14. ReliableXmit timeout
15. ReliableXmit timeout
16. ReliableXmit timeout
17. ReliableXmit timeout
(etc ... etc ...)
# sip show peer 103
* Name : 103
Secret :
MD5Secret :
Context : from-internal
Subscr.Cont. :
Language : fr
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox : 103 at default
VM Extension : 123
LastMsgsSent : 32767/65535
Call limit : 50
Dynamic : Yes
Callerid : "device" <103>
MaxCallBR : 384 kbps
Expire : 1229
Insecure : no
Nat : Always
ACL : Yes
T38 pt UDPTL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : XX.XX.XX.XX Port 7374
Defaddr->IP : 0.0.0.0 Port 5060
Transport : UDP
Def. Username: 103
SIP Options : (none)
Codecs : 0x10a (gsm|alaw|g729)
Codec Order : (g729:20,gsm:20,alaw:20)
Auto-Framing : No
100 on REG : No
Status : OK (140 ms)
Useragent : eyeBeam release 1102u stamp 52344
Reg. Contact : sip:103 at XXX:7374;rinstance=65465ec13147a5e9
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
Does anybody knows what is it ?
Thank you,
Simon
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(0111017) simonoch (reporter) - 2009-09-18 13:02
https://issues.asterisk.org/view.php?id=15890#c111017
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(sorry for my english)
Actually the plateform receive about 400 calls by a day, and the queue
generate a huge log.
So, how to filter this log?
In this time, the plateform is closed and we have :
# sip show channels
Peer User/ANR Call ID Format Hold
Last Message Expiry
XX.XX.XX.XX 103 247812e85e1a894 0x0 (nothing) No Init:
INVITE
XX.XX.XX.XX 103 48106940381414d 0x0 (nothing) No Init:
INVITE
XX.XX.XX.XX 103 6626309529c6fc6 0x0 (nothing) No Init:
INVITE
XX.XX.XX.XX 105 76b86216299928b 0x0 (nothing) No Init:
INVITE
4 active SIP dialogs
Simon
Issue History
Date Modified Username Field Change
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2009-09-18 13:02 simonoch Note Added: 0111017
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