check incominglimit and outgoinglimit in sip.conf http://www.voip-info.org
On 1/20/06, Kristian Larsson <[EMAIL PROTECTED]> wrote: > Hey! > > I'm having a small problem. I'm using Realtime to > store SIP account information. Dialing works just > fine, but when dialing a person already on the > phone I don't get a busy tone. > Eg, Phone 100 calls 200 and they chat with each other > phone 150 calls 100, and gets a regular ringing tone > > what I would is for phone 150 to receive a busy > tone since phone 100 is already speking with > someone else, how would I go about doing this? > > Kristian. > > _______________________________________________ > --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 > -- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org"; _______________________________________________ --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
