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?
