[asterisk-bugs] [Asterisk 0013935]: Calls are not beeing disconnected

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13935 
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Reported By:                kowalma
Assigned To:                
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Project:                    Asterisk
Issue ID:                   13935
Category:                   Channels/General
Reproducibility:            have not tried
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.0.1 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2008-11-19 14:43 CST
Last Modified:              2008-11-21 15:55 CST
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Summary:                    Calls are not beeing disconnected
Description: 
Hi,

I have very strange problem, that some calls remain in my asterisk box. I
use sesion timer so call should be dropped after expire time but it stays.

My main (I call it Asterisk-IBM) asterisk is 1.6.0.1 stable. Agents also
use same asterisk version. Each agent has installed asterisk on his PC.

On Asterisk-IBM I can see hanging calls ie:

asterisk-IBM*CLI> core show channel SIP/1127-afccc138
 -- General --LI>
           Name: SIP/1127-afccc138
           Type: SIP
       UniqueID: 1227122591.258108
      Caller ID: 1127
 Caller ID Name: agent--name
    DNID Digits: 0107040300050850522
       Language: en
          State: Ring (4)
          Rings: 0
  NativeFormats: 0x8 (alaw)
    WriteFormat: 0x8 (alaw)
     ReadFormat: 0x8 (alaw)
 WriteTranscode: No
  ReadTranscode: No
1st File Descriptor: 757
      Frames in: 2753
     Frames out: 1424
 Time to Hangup: 0
   Elapsed Time: 1h3m31s
  Direct Bridge: <none>
Indirect Bridge: <none>
 --   PBX   --
        Context: ccig
      Extension: 01070403000508505223
       Priority: 40
     Call Group: 0
   Pickup Group: 0
    Application: Dial
           Data: Local/1010505223 at mg
    Blocking in: ast_waitfor_nandfds
      Variables:
GOSUB_RETVAL=
gw=1010
typ=110
NoUseLCR=1
ProjektOK=1
BiuroOK=1
numer=505223
konsultant=000508
team=03
projektPOD=04
projektGL=07
projekt=0704
biuro=01
SPYGROUP=1127:000508
full=01070403000508505223
SIPCALLID=416663194bb91e142aab98333dadf0d4 at 192.168.1.127
SIPDOMAIN=sip.local
SIPURI=sip:1127 at 192.168.1.127

  CDR Variables:
level 1: godzina=20
level 1: licznik=1
level 1: clid="agent.name" <1127>
level 1: src=1127
level 1: dst=01070403000508505223
level 1: dcontext=ccig
level 1: channel=SIP/1127-afccc138
level 1: dstchannel=Local/1010505223 at mg-e1d6;1
level 1: lastapp=Dial
level 1: lastdata=Local/1010505223 at mg
level 1: start=2008-11-19 20:23:11
level 1: duration=3811
level 1: billsec=0
level 1: disposition=NO ANSWER
level 1: amaflags=DOCUMENTATION
level 1: uniqueid=1227122591.258108


when I login to agetnt's asterisk there is no call
CC1127*CLI> core show channels
Channel              Location             State   Application(Data)
0 active channels
0 active calls
584 calls processed
CC1127*CLI>

In Agent's CDR I can see that call had status "no answer"

CC1127:/var/log/asterisk/cdr-custom# grep 01070403000508505223 *
"","","konsultant","default","Dial","SIP/1127/01070403000508505223","2008-11-19
20:22:56","","2008-11-19 20:22:57","1","0","NO
ANSWER","DOCUMENTATION","","","1227122576.1055","","","","","","Local/konsultant at default-28d9;1"
CC1127:/var/log/asterisk/cdr-custom#

For today my Asterisk processed "133053 calls processed" and: 172 active
channels, 86 active calls hangs in system.


On my "master" Asterisk I have following setup for Session-Timer:
rtptimeout = 600
session-timers=accept
session-expires=120
session-minse=90
session-refresher=uas

rtptimeout=600

on client:
rtptimeout = 600
session-timers=accept
session-expires=120
session-minse=90
session-refresher=uac

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

---------------------------------------------------------------------- 
 (0095304) kowalma (reporter) - 2008-11-21 15:55
 http://bugs.digium.com/view.php?id=13935#c95304 
---------------------------------------------------------------------- 
I've just checked tcpdump and you are right - bye is missing. But I think
session timer should handle this?

I've attached screenshot from whireshark showing SIP between agent and IBM
box, and IBM box and asterisk acting as media gateway.

I've just checked, that this media-gateway asterisk is running 1.4.19-rc4,
so I will upgrade it tomorrow to 1.6.0.1 stable. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-11-21 15:55 kowalma        Note Added: 0095304                          
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