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 04:57:50.000-0600	Date Closed:	2006-12-05 12:57:13.000-0600
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Core/General
Versions:		Frequency of Occurrence
Related Issues:
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. ****** ADDITIONAL INFORMATION ****** extensions.conf [default] exten => 997960,1,Dial(SIP/084322997828@217.107.87.81|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! http://lists.digium.com/pipermail/svn-commits/2006-November/019440.html 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. Thanks. 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!