Hi, Thank you for the information. There are "t"s in Dial command in extensions.conf. When I deleted these "t"s, each sip phones were directly communicating. I just wrote these "t"s from the examples.
Does these "t" and "T" are used for transfer(blind/consaltation) from called user and calling user, respectively? If we don't have these 't' and 'T', can't we do transfer? Regards, Zen "Girish Gopinath" <[EMAIL PROTECTED]> wrote : > Zen, > > >I am trying to confirm the command 'canreinvite=yes' in sip.conf > >using grandstream BT101/2s and snom100s. In either case, no description > >nor 'canreinvite=yes', media stream always go through *. > > > >Do I need another settings for confirming sip clients directly > >communicate each other? > > Do you have a Dial statement that has "t" or "T" in it? > This will force the media stream to pass through Asterisk. > > Regards, Girish > > _________________________________________________________________ > Contact brides & grooms FREE! http://www.shaadi.com/ptnr.php?ptnr=hmltag > Only on www.shaadi.com. Register now! > > _______________________________________________ > 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 > _______________________________________________ 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
