Dear all i'm planning an upgrade of some asterisk installation from 1.4.32 to 1.6.0.28 (as i think it should be the most stable now).
Reading the UPGRADE-1.6.txt file i've noticed that: * SIP: 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. The "limitonpeer" configuration option is now renamed to "counteronpeer". As i've experienced some problem with 1.4 release about call-limit, i'd like to test this new counteronpeer functionality, but how to handle the ringinuse parmeter in queues.conf ? Basically i need that a sip user can make and receive more than one call (like a call-limit 3 setting) but i don't want that this interface rings if it is in a queue. Is it possible to do that? How? Thanks to all -- /*************/ nik600 http://www.kumbe.it -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
