[asterisk-bugs] [Asterisk 0011180]: call limits not work as expected (limitonpeer, busy-level)

[noreply at bugs.digium.com]

The following issue has been RESOLVED. 
====================================================================== 
http://bugs.digium.com/view.php?id=11180 
====================================================================== 
Reported By:                pj
Assigned To:                oej
====================================================================== 
Project:                    Asterisk
Issue ID:                   11180
Category:                   Channels/chan_sip/Subscriptions
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     resolved
Asterisk Version:            SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 89081 
Disclaimer on File?:        N/A 
Request Review:              
Resolution:                 fixed
Fixed in Version:           
====================================================================== 
Date Submitted:             11-07-2007 05:05 CST
Last Modified:              11-19-2007 03:12 CST
====================================================================== 
Summary:                    call limits not work as expected (limitonpeer,
busy-level)
Description: 
probably most wanted scenario is to let users place unlimited number of
outgoing calls from their sip phones (to be able to transfer etc.), but
indicate busy condition for calls to their phones, when one call already
exist (no matter if incomming or outgoing). 
to achieve this, I set 'limitonpeer=yes' in [general] to be able to limit
only outgoing calls from asterisk to sip device (peer from asterisk
perspective, incomming call from sip phone perspective) and set
call-limit=1 in type=friend phone definition.
but seems, that limitonpeer=yes is actualy not working, because from sip
phone I can place only single call, so limited is also 'user' part in
asterisk phone definition.
I tried also set call-limit=2 and busy-level=1, but this permit call to
sip phone, even if I have already placed call from this phone. I think that
'busy-level' should work or count calls fot both directions, to correctly
indicate 'busy' condition if user already has call.
====================================================================== 

---------------------------------------------------------------------- 
 oej - 11-19-07 03:12  
---------------------------------------------------------------------- 
Fixed in svn trunk, rev 89406 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
11-19-07 03:12  oej            Status                   new => resolved     
11-19-07 03:12  oej            Resolution               open => fixed       
11-19-07 03:12  oej            Assigned To               => oej             
11-19-07 03:12  oej            Note Added: 0073906                          
======================================================================