On Monday 15 March 2004 16:00, Olle E. Johansson wrote: > > It depends on your SIP device. Asterisk places the call to your SIP device > regardless, since by SIP protocol design the UA is not a "slave", it is > free. So the SIP ua must answer "busy" for Asterisk to understand that > you're busy. If not, the call is placed to you and Asterisk has no > knowledge that you are busy. Check you SIp phone if you can limit the > number of concurrent calls.
So does anyone know if the Grandstream handytone-286 sends this "busy" answer ? I'm guessing it doesn't. In that case, what other ways are there of connecting my dect phones to a voip * system ? - can I just connect them into the x100p's phone socket (how do I send calls to that port) or do I need to get a fxs card and run wire's everywhere - not an option :( How does everyone else connect up DECT phones to a * based system. Surely * should know if a phone is in use ? After all it initiated/took part in the call in the first place ;) Jon _______________________________________________ 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
