Thank you this does it in SIPDefault.cnf: # Allow for the bridge on a 3way call to join remaining parties upon hangup cnf_join_enable: 1; 0-Disabled, 1-Enabled (default)
On 11/14/05, C F <[EMAIL PROTECTED]> wrote: > Thank you Chris, I will look into it and report back. > > On 11/14/05, Chris Wade <[EMAIL PROTECTED]> wrote: > > C F wrote: > > > Cisco 7960 gets a call from zap/1, hits conf to call out on zap/2, > > > then hits join, after a while cisco hangsup, at which point zap/1 and > > > zap/2 can still talk, shouldn't asterisk hangup on all three? > > > > I'll assume SIP here since SCCP conf is a work in progress. Under SIP, > > there is a .cnf option in SIPDefault.cnf / SIPMAC.cnf that allows you to > > specify if the calls should be re-invited to each other (allowing those > > calls to continue) or if they should be hung-up. I've completely > > switched my network over to chan_sccp so I don't even have a copy of my > > SIPDefault.cnf anymore, otherwise I would send you a copy with all > > available options detailed. > > > > -- > > Christopher L. Wade, CCNA, CCDA, CQS-CIPTES, CQS-CWLSS > > > > _______________________________________________ > > --Bandwidth and Colocation sponsored by Easynews.com -- > > > > Asterisk-Users mailing list > > [email protected] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
