[Asterisk-Users] sip ata's

[vampares]

   Hello.  I have a linux and two sip-ata's, a sipura 2002 and a GS ht-386.  I 
also have three sipphone numbers.  I can connect the atas to the sipphone 
accounts and I get a dial tone and I can call my house and it says, "Thank 
you for using SipPhone..."
   Using asterisk, I have the ata's registering to my computer and I register 
two sipphone numbers with my computer.  When I pick up the phone I don't get 
a dialtone.  I can use kphone and call a sipphone and the logs come back 
saying I have phone on hook, phone is off the hook, and one phone rings 
usually, one comes back busy (in log).  I pick-up the phone and nobody is 
there and then the asterisk-voicemail kicks in.

   I guess I have two questions:
Where is the dial-tone?  I noticed I compiled "phone sounds" but my ata has a 
dial-tone when its not serviced.

My grandstream 386 has 2 fxs's.  One of them clicks on and off and on and off 
when I pick up the receiver even though it rings when I call it.  I have it 
set up the same as the other port as best as I can.  I think it may be a 
setting on the 386 that I'm not seeing.  Is there anyone aware of what causes 
this?

I also noticed that when the call is handled by asterisk there is an invite.  
Is this a reinvite and where do the canreinvite/reinvites go?