[asterisk-users] Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)

[Vieri]

--- "Nestor A. Diaz" <nestor at tiendalinux.com> wrote:

> Mojo with Horan & Company, LLC wrote:
> > Nestor A. Diaz wrote:
> >   
> >> 1. I use a queue with just on sip device, one
> call at a time, however 
> >> and without reason just after some couple of
> hours the sip device show 
> >> in use and then no calls are transfered from the
> queue to the sip 
> >> device, i do a sip show inuse and this is the
> result:asterisk -rx "sip 
> >> show inuse"
> >> * User name               In use          Limit
> >> 200                     0               3
> >> * Peer name               In use          Limit
> >> 200                     1/0             3

Did you try a "show channels" to see if there were
stale channels for peer 200?

I had the same problem you describe but it was due to
"hung" channels (used * 1.4.18.1 with rtp*timeout and
saw "inuse" peers during the pre-timeout periods even
though the agents weren't on a call).



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