In which future release of Asterisk are we (since it is open-source, we theoretically have "some" control) going to stop renaming and deprecating features?
-----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Gareth Blades Sent: Friday, May 07, 2010 8:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Getting presence working in 1.6.2 Richard Kenner wrote: >> I read the wiki and see mention about needing to set call-limit in >> asterisk 1.4 but that has been depreciated in 1.6 so what is the way it >> should be done in 1.6? > > I set > > callcounter=yes > > in sip.conf. > Thanks that works perfectly. -- _____________________________________________________________________ -- 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 -- _____________________________________________________________________ -- 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
