it sounds like a SIP BYE message might not be sent.
How can it be checked/solved?
Sip debugging and wireshark will help.
perhaps a little more detail would help
Ian
Sorry this post was the follow ip of this topic http://forums.digium.com/viewtopic.php?t=68748&highlight=.
As a resume I have asterisk 1.6.0.9 with only a few SIP extensions (no other functionnalities yet) and an ATA (Grandstream Handytone 502). When a call is established, the hangup is never detected by the devices connected to the ATA.
The configuration has been made on Ubuntu Server with asterisk-gui 2
I would really appreciate if anyone could help me finding the solution because I have this trouble for months…
Thanks in advance