[Asterisk-Users] Problem with Cisco local conference and hangup

C F shmaltz at gmail.com
Mon Nov 14 15:17:59 MST 2005


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 <shmaltz at gmail.com> wrote:
> Thank  you Chris, I will look into it and report back.
>
> On 11/14/05, Chris Wade <clwade at sparco.com> 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
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>



More information about the asterisk-users mailing list