Summary:	ASTERISK-10818: problems with counting call limits
Reporter:	pj (pj)	Labels:
Date Opened:	2007-11-19 12:56:09.000-0600	Date Closed:	2008-01-09 12:20:55.000-0600
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Channels/chan_sip/General
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	when I have one call FROM sip phone and make another concurent call TO phone, 'sip show inuse' displays: * Peer name In use Limit 324 2/1/0 2 but after hangup it displays some weird values, even if phone s idle, like: * Peer name In use Limit 324 -1/0/0 2 it can be repaired, only with restarting asterisk second issue: - if I make call FROM sip phone (ie. asterisks friend user part), 'sip show inuse' still shows zero in 'in use' column of 'user' configuration part, * User name In use Limit 324 0 2 instead, calls are counted only in 'peer' section, when I make call FROM phone, it shows: * Peer name In use Limit 324 1/0/0 2 I have 'limitonpeer=yes' in sip.conf, to apply limits only to peer part, as in sip.conf.sample limitonpeer = yes              ; Apply call limits on peers only. third issue: - 'limit 2' is still displayed in 'user' part, even if I have 'limitonpeer=yes' in sip.conf ****** ADDITIONAL INFORMATION ****** [general] limitonpeer=yes [324] type=friend call-limit=2 busy-level=1
Comments:	By: Olle Johansson (oej) 2007-11-27 01:27:13.000-0600 The third issue is something I consider cosmetic. To add the required processing to find out which users that actually has matching peers and remove them from the output is just extra processing which might cause more locking issues than you need. The second issue doesn't seem like a bug when you have limitonpeer=yes. If you have that, we only use the call counter on the peer half of a friend and the user counter is never used. The first issue seems like a bug, but there are several related bug reports here in the bug tracker. I think this is a duplicate bug report. By: Digium Subversion (svnbot) 2007-11-27 01:31:48.000-0600 Repository: asterisk Revision: 89624 U   branches/1.4/configs/sip.conf.sample ------------------------------------------------------------------------ r89624 | oej | 2007-11-27 01:31:47 -0600 (Tue, 27 Nov 2007) | 6 lines Clarify limitonpeers=yes (closes issue ASTERISK-10818) Reported by: pj ------------------------------------------------------------------------ By: Digium Subversion (svnbot) 2007-11-27 01:34:23.000-0600 Repository: asterisk Revision: 89625 _U  trunk/ U   trunk/configs/sip.conf.sample ------------------------------------------------------------------------ r89625 | oej | 2007-11-27 01:34:22 -0600 (Tue, 27 Nov 2007) | 14 lines Merged revisions 89624 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89624 | oej | 2007-11-27 08:34:19 +0100 (Tis, 27 Nov 2007) | 6 lines Clarify limitonpeers=yes (closes issue ASTERISK-10818) Reported by: pj ........ ------------------------------------------------------------------------ By: Digium Subversion (svnbot) 2008-01-08 14:38:44.000-0600 Repository: asterisk Revision: 97192 U   branches/1.4/apps/app_voicemail.c ------------------------------------------------------------------------ r97192 | mmichelson | 2008-01-08 14:38:44 -0600 (Tue, 08 Jan 2008) | 9 lines Making some changes designed to not allow for a corrupted mailstream for a vm_state. 1. Add locking to the vm_state retrieval functions so that no linked list corruption occurs. 2. Make sure to always grab the persistent vm_state when mailstream access is necessary. 3. Correct an incorrect return value in the init_mailstream function. (closes issue ASTERISK-10818, reported by dwhite) ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=97192 By: Digium Subversion (svnbot) 2008-01-08 14:42:10.000-0600 Repository: asterisk Revision: 97193 _U  trunk/ U   trunk/apps/app_voicemail.c ------------------------------------------------------------------------ r97193 | mmichelson | 2008-01-08 14:42:10 -0600 (Tue, 08 Jan 2008) | 17 lines Merged revisions 97192 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97192 | mmichelson | 2008-01-08 14:42:07 -0600 (Tue, 08 Jan 2008) | 9 lines Making some changes designed to not allow for a corrupted mailstream for a vm_state. 1. Add locking to the vm_state retrieval functions so that no linked list corruption occurs. 2. Make sure to always grab the persistent vm_state when mailstream access is necessary. 3. Correct an incorrect return value in the init_mailstream function. (closes issue ASTERISK-10818, reported by dwhite) ........ ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=97193 By: Digium Subversion (svnbot) 2008-01-09 12:20:55.000-0600 Repository: asterisk Revision: 97535 _U  team/murf/bug11210/ U   team/murf/bug11210/CHANGES U   team/murf/bug11210/apps/app_queue.c U   team/murf/bug11210/apps/app_voicemail.c U   team/murf/bug11210/channels/chan_console.c U   team/murf/bug11210/channels/chan_mgcp.c U   team/murf/bug11210/channels/chan_sip.c U   team/murf/bug11210/configs/queues.conf.sample U   team/murf/bug11210/funcs/func_groupcount.c U   team/murf/bug11210/main/asterisk.c U   team/murf/bug11210/main/autoservice.c U   team/murf/bug11210/main/utils.c U   team/murf/bug11210/res/res_jabber.c ------------------------------------------------------------------------ r97535 | murf | 2008-01-09 12:20:51 -0600 (Wed, 09 Jan 2008) | 140 lines Merged revisions 97125,97153-97154,97193,97196-97200,97203,97208,97248 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r97125 | tilghman | 2008-01-08 12:06:27 -0700 (Tue, 08 Jan 2008) | 11 lines Merged revisions 97077 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97077 | tilghman | 2008-01-08 12:02:13 -0600 (Tue, 08 Jan 2008) | 3 lines Apply multiple crash fixes, found in issue ASTERISK-10897, but not completely closing that issue. ........ ................ r97153 | file | 2008-01-08 12:55:56 -0700 (Tue, 08 Jan 2008) | 12 lines Merged revisions 97152 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97152 | file | 2008-01-08 15:53:52 -0400 (Tue, 08 Jan 2008) | 4 lines If no group has been provided to the GROUP_COUNT dialplan function then use the first one specific to the channel. (closes issue ASTERISK-10610) Reported by: m4him ........ ................ r97154 | file | 2008-01-08 13:06:52 -0700 (Tue, 08 Jan 2008) | 6 lines Move common code for setting T38 capabilities and fix a bug with fax detection in the SIP RTP read callback. It's still sort of silly... but more on that later. (closes issue ASTERISK-10758) Reported by: dimas Patches:      sipt38prop.patch uploaded by dimas (license 88) ................ r97193 | mmichelson | 2008-01-08 13:45:59 -0700 (Tue, 08 Jan 2008) | 17 lines Merged revisions 97192 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97192 | mmichelson | 2008-01-08 14:42:07 -0600 (Tue, 08 Jan 2008) | 9 lines Making some changes designed to not allow for a corrupted mailstream for a vm_state. 1. Add locking to the vm_state retrieval functions so that no linked list corruption occurs. 2. Make sure to always grab the persistent vm_state when mailstream access is necessary. 3. Correct an incorrect return value in the init_mailstream function. (closes issue ASTERISK-10818, reported by dwhite) ........ ................ r97196 | file | 2008-01-08 13:50:57 -0700 (Tue, 08 Jan 2008) | 14 lines Merged revisions 97195 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97195 | file | 2008-01-08 16:48:20 -0400 (Tue, 08 Jan 2008) | 6 lines Fix various DTMF issues in chan_mgcp. (closes issue ASTERISK-10954) Reported by: eferro Patches:      dtmf_control_hybrid-inband-mode.patch uploaded by eferro (license 337) ........ ................ r97197 | file | 2008-01-08 13:52:07 -0700 (Tue, 08 Jan 2008) | 2 lines One line documentation ftw! ................ r97198 | tilghman | 2008-01-08 13:56:38 -0700 (Tue, 08 Jan 2008) | 11 lines Merged revisions 97194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97194 | tilghman | 2008-01-08 14:47:07 -0600 (Tue, 08 Jan 2008) | 3 lines Increase constants to where we're less likely to hit them while debugging. (Closes issue ASTERISK-11165) ........ ................ r97199 | oej | 2008-01-08 13:58:56 -0700 (Tue, 08 Jan 2008) | 2 lines Iksemel is alive in a new home. Release 1.3 is out with bug fixes. ................ r97200 | oej | 2008-01-08 14:01:46 -0700 (Tue, 08 Jan 2008) | 4 lines Change reference to external library so it appears on the extref listing http://www.asterisk.org/doxygen/trunk/extref.html ................ r97203 | mmichelson | 2008-01-08 14:18:32 -0700 (Tue, 08 Jan 2008) | 8 lines Adding the option of specifying a second interface in a member definition for a queue. app_queue will monitor this second device's state for the member, even though it actually calls the first interface. This ability has been added for statically defined queue members, realtime queue members, and dynamic queue members added through the CLI, dialplan, or manager. (closes issue ASTERISK-11085, reported by acidv) ................ r97208 | mmichelson | 2008-01-08 14:28:04 -0700 (Tue, 08 Jan 2008) | 10 lines Blocked revisions 97206 via svnmerge ........ r97206 | mmichelson | 2008-01-08 15:24:48 -0600 (Tue, 08 Jan 2008) | 3 lines Some coding guidelines-related cleanup ........ ................ r97248 | twilson | 2008-01-08 14:56:57 -0700 (Tue, 08 Jan 2008) | 2 lines Initialize new variable to NULL ................ ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=97535