[asterisk-bugs] [Asterisk 0015890]: 1.6.1.5 - "Ghost" channels

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=15890 
====================================================================== 
Reported By:                simonoch
Assigned To:                
====================================================================== 
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:              
====================================================================== 
Date Submitted:             2009-09-14 07:41 CDT
Last Modified:              2009-09-18 13:05 CDT
====================================================================== 
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

====================================================================== 

---------------------------------------------------------------------- 
 (0111018) simonoch (reporter) - 2009-09-18 13:05
 https://issues.asterisk.org/view.php?id=15890#c111018 
---------------------------------------------------------------------- 
#sip show channel 247812e85e1a894

  * SIP Call
  Curr. trans. direction:  Outgoing
  Call-ID:                247812e85e1a8942323c94ba18143aaa at astserver
  Owner channel ID:       
  Our Codec Capability:   266
  Non-Codec Capability (DTMF):   1
  Their Codec Capability:   0
  Joint Codec Capability:   266
  Format:                 0x0 (nothing)
  T.38 support            No
  Video support           No
  MaxCallBR:              384 kbps
  Theoretical Address:    82.241.145.220:11198
  Received Address:       82.241.145.220:11198
  SIP Transfer mode:      open
  NAT Support:            Always
  Audio IP:               astserver (local)
  Our Tag:                as62ecb29f
  Their Tag:              
  SIP User agent:         
  Username:               103
  Peername:               103
  Original uri:          
sip:103 at XX.XX.XX.XX:11198;rinstance=c452a8553eb4835b
  Need Destroy:           No
  Last Message:           Init: INVITE
  Promiscuous Redir:      No
  Route:                  N/A
  DTMF Mode:              rfc2833
  SIP Options:            (none)
  Session-Timer:          Inactive 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-09-18 13:05 simonoch       Note Added: 0111018                          
======================================================================