Summary:ASTERISK-03502: [patch] Improve call limit handling
Reporter:Olle Johansson (oej)Labels:
Date Opened:2005-02-13 12:25:06.000-0600Date Closed:2008-01-15 15:25:15.000-0600
Versions:Frequency of
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


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)