[Asterisk-Users] SIP channel not diconnecting on hangup

[Scott Bussinger]

I've got an SPA-841 SIP hardphone connecting to my asterisk server across 
the internet through a couple of NAT routers. Everything works great (I can 
initiate calls, receive calls, hear audio both ways, etc.) except for one 
thing. When I hang up the phone, the connection in asterisk doesn't 
disconnect. The phone is idle and things everything is fine, but Asterisk 
still show an open channel. It's like the phone isn't sending some sort of 
disconnect message to Asterisk. Can anyone provide some ideas on what might 
be going wrong?

As a test case, I call my echo() extension from the remote phone. The 
connection works fine but when I hangup the phone and get information from 
the Asterisk console here's what I see:

[Jan 27 10:27:00]     -- Executing Playback("SIP/scottbhome-f4de", 
"demo-echotest") in new stack
[Jan 27 10:27:00]     -- Playing 'demo-echotest' (language 'en')
[Jan 27 10:27:19]     -- Executing Echo("SIP/scottbhome-f4de", "") in new 
stack

**** I hangup the phone here ****

pbx*CLI> show channels
Channel              Location             State   Application(Data)
SIP/scottbhome-f4de  6300 at internal:2      Up      Echo()
1 active channels
1 active calls

pbx*CLI> sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Form  Hold     Last 
Message
xx.xx.xx.xx    scottbhome  304dcbc8-5f  00101/00102  g729  No       Rx: 
INVITE
1 active SIP channels


So the connection initiates correctly, but nothing ever terminates it. I 
finally do a SOFT HANGUP to kill the connection.

Thanks for any help!