Summary:	ASTERISK-03502: [patch] Improve call limit handling
Reporter:	Olle Johansson (oej)	Labels:
Date Opened:	2005-02-13 12:25:06.000-0600	Date Closed:	2008-01-15 15:25:15.000-0600
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Core/General
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:	( 0) limits.txt
Description:	* do not lookup user if there's no call limits  (Removes one SQL lookup for realtime db) * Respond properly when call limit is reached (SIP error) * Fix "sip show inuse" to show peers as well * "sip show inuse" now only lists devices with limits  "sip show inuse all" lists all (current behaviour) Disclaimer on file ****** ADDITIONAL INFORMATION ****** I use "480 Temporarily unavailable" as SIP error when we reject an incoming call due to call limit.
Comments:	By: Kevin P. Fleming (kpfleming) 2005-02-13 12:29:08.000-0600 I thought this "call limit" stuff in chan_sip was deprecated? By: Olle Johansson (oej) 2005-02-13 12:32:55.000-0600 Well, outgoing limit is something I've been trying to get rid of many times, since it clearly does not work... But Mark wants to fix it instead of getting rid of it, so I will take a look at it. Meanwhile, I'm trying to update [peer]s to support all that [user]s do. Hint, hint. By: Mark Spencer (markster) 2005-02-13 12:41:11.000-0600 Added to CVS, thanks! By: Russell Bryant (russell) 2005-02-13 23:26:00.000-0600 not included in 1.0 By: Digium Subversion (svnbot) 2008-01-15 15:25:15.000-0600 Repository: asterisk Revision: 5021 U   trunk/channels/chan_sip.c ------------------------------------------------------------------------ r5021 | markster | 2008-01-15 15:25:15 -0600 (Tue, 15 Jan 2008) | 2 lines Merge limits patch (bug ASTERISK-3502) ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=5021