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Summary:ASTERISK-01025: SIP inUse counter fails to decrease
Reporter:z_smurf (z_smurf)Labels:
Date Opened:2004-02-11 16:55:44.000-0600Date Closed:2011-06-07 14:10:39
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
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Description:In some circumstances, the inUse counter fails to decrease back to 0 when no sip channels longer exists.
It has happend tree times now with CVS 2004-01-20 and 2004-02-11. I believe it happens when someone uses several sip-phones with exact the same config, and just pulls the network-plug from one to another.

****** ADDITIONAL INFORMATION ******

To zero the inUse-counter I have to restart asterisk.

I have added a lot of debug-output to try find where it goes wrong, but if anybody has an idea whats up, please add a note. An Idea I have is that the function to decrease the inUse-counter find_user() does not find the user because the phone's IP has changed? But I cannot find anything about that in the logs...
Comments:By: Brian West (bkw918) 2004-02-11 23:21:03.000-0600

known issue bug number 207, if you can do a sip show channels and see if the channel is still hung if so then 207 needs to be implemented ASAP.

bkw
By: z_smurf (z_smurf) 2004-02-12 17:09:59.000-0600

No,
"sip show channels" shows 0 channels.
"show channels" shows 0 channels.
"sip show inuse" shows 1 inuse on one peer.

Do you think a channel really is open? I think there is no channel, just the counter is wrong.
By: zoa (zoa) 2004-02-14 08:52:37.000-0600

I think i can confirm something similar

agent108        0               N/A             0               N/A
agent107        0               N/A             0               N/A
agent106        23              N/A             0               N/A
agent105        0               N/A             0               N/A
agent104        0               N/A             0               N/A
agent103        0               N/A             0               N/A

but the server is with an old cvs version.


scxgk3*CLI> sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Lag      Jitter  Format
10.0.1.47        agent242    CCBBC11F-A8  00101/42061  00000ms  0000ms  ILBC
10.0.1.150       (None)      EB18400C-52  00101/44851  00000ms  0000ms  UNKN
By: bfranks (bfranks) 2004-02-25 16:55:55.000-0600

This occurs on my Asterisk CVS-01/18/04-12:30:42 on very rare occasions.  I simply issue a restart gracefully to resolve it.

- Brent
By: Olle Johansson (oej) 2004-02-26 15:34:11.000-0600

THe incominglimit/outgoinglimit stuff need to be revised in detail. Right now the outgoinglimit is disabled in source and it seems like the incominglimit is not correct. If we don't have anyone coming forward here, we need to decide if this half-implementation should be ripped out for 1.0 release. As it stands now, it's not very well done.
By: pliew (pliew) 2004-03-10 21:19:32.000-0600

The incominglimit decrements correctly assuming comms between sip ua and * works cleanly - which is most of the time. I've had problems on a couple of occasions, where the GS phone (I know its not the best) drops a call when a user takes the call of handsfree and back on again (and maybe some other scenario). The result of this dropped call is not being picked up by the incominglimit code - need to do some testing to see what codes get returned - currently only 487 (channel destroyed) is being monitored. Any help here appreciated.
By: Brian West (bkw918) 2004-04-17 23:48:29

Please comment if this is still a problem and the bug will go into feedback.

Thanks,
House Keeping