Hi,
Terry H. Gilsenan wrote:
I was having this problem with Gradstream BT101's with Asterisk @ Home
version 0.7.
The problem was that there was a sip channel still open (as far as asterisk
and the phone were concerned) however this sip channel was not actually in
use. The existence of this sip channel meant that whilst the phone could
make calls, any incoming calls were directed to voicemail.
Thanks for the hint. I did control the channels, they were all closed but the
problem was still there.
After testing the meeting app though (calling in via a PSTN->Cisco->Asterisk
there is indeed a hung channel. Anyone knows what could be causing this?
--
Channel (Context Extension Pri ) State Appl. Data
Zap/pseudo-1655835607 (default s 1 ) Rsrvd (None)
(None)
SIP/x.x.x.x-0814dbb8 (la-in 310xxxxxxxx 2 ) Up MeetMe
|ip
2 active channel(s)
--
cheers,
Arnd
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