Hi again, I think I have now a workaround for call transfer on ATA 186.
This is the extension corresponding to the phone connected to an ATA186 exten => 103,1,Dial(SIP/103,20),Tt exten => 103,2,Voicemail2(us101) exten => 103,3,Hangup exten => 103,102,Ringing exten => 103,103,Wait(1) exten => 103,104,Goto(1) I can now to attended transfer a call to this phone too. The strange thins is that if I call this extension when the phone in off-hook but not in a call, it rings for 1 second then exit with a busy tone. Why? Thanks, Dan ----- Original Message ----- From: "Dan" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 30, 2003 5:44 PM Subject: Re: [Asterisk-Users] Dummy account/extension > Hi, > > Thanks for the suggestion. > I have change it like that: > > ;dummy extension > exten => 199,1,Ringing > exten => 199,2,Wait(60) ; give illusion we might pick up > exten => 199,3,Hangup > > in order to hear the ring too. > > ..but now... how can I do to call this extension from a Dial command? > > What I want in the final is to have a workaround for ATA186 in order to > prevent consider it busy during the attended transfer. > More, I want to prevent been bussy when not in a call. The Call Waiting does > not function during the dialtone period, just during the call. > > There is any other way to do it? > > Thanks for your help, > Dan > > > > > ----- Original Message ----- > From: "Armand A. Verstappen" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Wednesday, July 30, 2003 5:06 PM > Subject: Re: [Asterisk-Users] Dummy account/extension > > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
