[Asterisk-Users] Dial to a SIP fone ends up at Voicemail Busy

[Arnd Vehling]

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