SVN commits to the Digium repositories wrote: > Author: oej > Date: Sun Nov 25 05:46:17 2007 > New Revision: 89554 > > URL: http://svn.digium.com/view/asterisk?view=rev&rev=89554 > Log: > - Deprecate "call-limit" in chan_sip. No other channel driver enforces > call-limits > and we now have the groupcount system to implement call-limits in the > dialplan. You > can use the "setvar" option in realtime/sip.conf to set limits per device. > > - Implement "callcounter" as a new option to enable the call counting we need > to > report device status to queue, manager and SIP subscriptions. > > The call counter setting is now enabled in the code by setting the device > call-limit > to 999. When we remove the call limit, we can simply enable this with a > boolean > setting. > > Modified: > trunk/CHANGES > trunk/channels/chan_sip.c > trunk/configs/sip.conf.sample > > Modified: trunk/CHANGES > URL: > http://svn.digium.com/view/asterisk/trunk/CHANGES?view=diff&rev=89554&r1=89553&r2=89554 > ============================================================================== > --- trunk/CHANGES (original) > +++ trunk/CHANGES Sun Nov 25 05:46:17 2007 > @@ -89,8 +89,14 @@ > * SIP now adds a header to the CANCEL if the call was answered by another > phone > in the same dial command, or if the new c option in dial() is used. > * The new default is that 100 Trying is not sent on REGISTER attempts as > the RFC specifically > - states it is not needed. For phones, however, that do require it the > registertrying option > + states it is not needed. For phones, however, that do require it the > "registertrying" option > has been added so it can be enabled. > + * The "call-limit" option is marked as deprecated. It still works in this > version of > + Asterisk, but will be removed in the following version. Please use the > groupcount functions > + in the dialplan to enforce call limits. > + * A new option called "callcounter" (global/peer/user level) enables call > counters needed > + for better status reports needed for queues and SIP subscriptions. > (Call-Limit was previously > + used to enable this functionality). >
The first part of this that talks about deprecating "call-limit" should go in UPGRADE.txt instead of CHANGES. CHANGES - a list of new features UPGRADE.txt - All of the information about deprecated features, syntax changes, and other changes in behavior that users need to know when upgrading to this major version from the previous one. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
