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

[Mojo with Horan & Company, LLC]

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
>
> Simple workaround: delete sip 200 entry from sip.conf, reload sip.conf, 
> recreate 200 extensions and reload sip.conf
>   
Does a simple sip reload work, or do you really need to go to all the 
trouble of removing the peer definition?