Summary:ASTERISK-08185: Called SIP subscriber still ringing even after hanging up the call by calling side
Reporter:Pavel Yarmolchuk (xpasha)Labels:
Date Opened:2006-11-25 04:57:50.000-0600Date Closed:2006-12-05 12:57:13.000-0600
Versions:Frequency of
Environment:Attachments:( 0) sipdebug.txt
Description:Called SIP subscriber still ringing even after canceling the call by calling side. The same situation with 1.4-beta3.



exten => 997960,1,Dial(SIP/084322997828@|10)

the call comes to 997960 from PSTN over Cisco AS5350 with and go to the other Cisco AS5350.
Comments:By: Steve Murphy (murf) 2006-11-25 10:31:51.000-0600

Hmmm. I wonder if this could be a reincarnation of 7433, where devicestate is not being updated? If you want, you can set the call_limit on that peer, and see if that changes anything...!

By: Pavel Yarmolchuk (xpasha) 2006-11-25 17:41:57.000-0600

Nothing changed with call-limit on this SIP peers, anyway this is not good workaround. I wonder how this bug still present and nobody tell about it.

By: Steve Murphy (murf) 2006-11-25 23:06:37.000-0600

Hmm. Thanks for testing. It was a shot in the dark, I admit.

By: Joshua C. Colp (jcolp) 2006-11-29 21:19:37.000-0600

Assigned to oej. This is the CANCEL/BYE bug that some people have brought up I do believe.

By: Olle Johansson (oej) 2006-11-30 01:14:01.000-0600

This is definitely a duplicate bug report.

By: Serge Vecher (serge-v) 2006-11-30 13:03:00.000-0600

xPasha: please try out the invitestate-1.4 branch that oej has created to address this issue and provide some feedback here please. Thanks!


By: Pavel Yarmolchuk (xpasha) 2006-12-01 07:39:31.000-0600

*CLI> core show version
Asterisk SVN-oej-invitestate-1.4-r48132 built by root @ obzvonki on a i686 running Linux on 2006-12-01 13:35:53 UTC

This problem still present. :(

By: Serge Vecher (serge-v) 2006-12-01 11:06:10.000-0600

alright, let's see the updated debug log then.

By: Olle Johansson (oej) 2006-12-01 11:31:18.000-0600

Please enable sip history and dumphistory in sip.conf and run "sip show channels" after the call. If there's still an open sip channel, run "sip show history" on that channel.


By: hristo (hristo) 2006-12-02 12:30:12.000-0600

I have just added a note and two sip debug files for ivitestate-1.4 branch to issue ASTERISK-8038095, which describes the same problem.
I even get different behavior depending on the time the cancel is sent - after trying ot after 183 message.

By: Olle Johansson (oej) 2006-12-02 14:34:50.000-0600

This was related to 183 Session progress.

Please test latest version of the branch of your choice
- invitestate for 1.2
- invitestate-1.4 for 1.4
- trunk for trunk

I believe this is fixed. Your urgent reply is really appreciated, so we can close this issue and merge
the branches. Thanks.

Many thanks to Leif Madsen (Blitzrage) who gave me access to a test system and for placing
many test calls. International bug hunting - I'm in Sweden, Leif's in Canada and the server was
located in Florida!

By: Pavel Yarmolchuk (xpasha) 2006-12-04 02:19:51.000-0600

For invitestate-1.4 this problem no longer present. Thanks!

By: Olle Johansson (oej) 2006-12-05 12:57:00.000-0600

resolved in asterisk 1.4 svn.

Thanks for testing quickly!