Ok thanks, do you think that it isn't possible to do that automatically from asterisk?
On 12/12/06, Pavel Jezek <[EMAIL PROTECTED]> wrote:
I think, that adhoc conferencing isn't possible in this way, instead you should use meetme, ie.: skinny user calls to user A and transfer his to meetme number skinny user calls to user B and transfer his to meetme number skinny user calls to meetme number all three speech in conference... nik600 wrote: > Hi, can i set up my asterisk for: > > - receive a skinny call in a specific context (yes, i have already > compiled asteirsk with h323 support) > - forward the call to a sip user A > - make the sip user B join the call and create a conference between > skinny caller, A and B > > maky thanks > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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