Summary:	ASTERISK-10617: Hint stuck on hold after attended transfer with notifyhold=yes in sip.conf
Reporter:	Francesco Romano (francesco_r)	Labels:
Date Opened:	2007-10-25 07:31:00	Date Closed:	2007-11-05 12:50:23.000-0600
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Channels/chan_sip/General
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:	( 0) verbosedebug.txt
Description:	This is the show hints with 0 calls: 202@ext-local           : SIP/202               State:Hold            Watchers  0 I have attached a full debug with sip history.
Comments:	By: Joshua C. Colp (jcolp) 2007-10-31 20:11:20 Can you please provide step by step the flow of calls so I can reproduce this? Thanks. By: Francesco Romano (francesco_r) 2007-11-01 09:38:33 The log i have attached it's this call flow: - extension 201 (Grandstream Gxp2000) call extension 202 (Grandstream BT200) - 202 flash and call an IAX trunk - after the answer 202 do an attended transfer - 202 stuck on hold after the hangup This is 100% reproducible. By: Badalian Vyacheslav (slavon) 2007-11-02 02:24:18 I also have this problem. Have Cisco 7940 (ex. 850) that member of queue. 1. Call from queue (member 850). 2. Answer. (member state in queue - In Use) 3. Attended transfer (Cisco menu->More->Transfer) from 111. 4. Answer (first call go to hold) (member state in queue - On Hold) 5. Hangup second call. (member state in queue - On Hold) 6. Unhold second call (member state in queue - On Hold - WORNG) 7. Hangup first call (member state in queue - On Hold - WORNG) Result - member not get any calls from queue because it "On Hold". "sip reload" and "reload app_queue" not help to us for unhold it. Only "stop now", asterisk -gT I can't provide sip debug information because have much voice traffic. Now time i turn off notifyhold=yes. By: Leif Madsen (lmadsen) 2007-11-05 12:16:18.000-0600 Reproduced!  Joshua is now adding in some debugging information. More to follow. By: Digium Subversion (svnbot) 2007-11-05 12:45:56.000-0600 Repository: asterisk Revision: 88671 U   branches/1.4/channels/chan_sip.c ------------------------------------------------------------------------ r88671 | file | 2007-11-05 12:45:55 -0600 (Mon, 05 Nov 2007) | 7 lines If a SIP channel is put on hold multiple times do not keep incrementing the onHold value. (closes issue ASTERISK-10617) Reported by: francesco_r Tested by: blitzrage (closes issue ASTERISK-10099) Reported by: acennami ------------------------------------------------------------------------ By: Digium Subversion (svnbot) 2007-11-05 12:50:23.000-0600 Repository: asterisk Revision: 88673 _U  trunk/ U   trunk/channels/chan_sip.c ------------------------------------------------------------------------ r88673 | file | 2007-11-05 12:50:22 -0600 (Mon, 05 Nov 2007) | 15 lines Merged revisions 88671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88671 | file | 2007-11-05 14:47:13 -0400 (Mon, 05 Nov 2007) | 7 lines If a SIP channel is put on hold multiple times do not keep incrementing the onHold value. (closes issue ASTERISK-10617) Reported by: francesco_r Tested by: blitzrage (closes issue ASTERISK-10099) Reported by: acennami ........ ------------------------------------------------------------------------