[asterisk-users] Queues and Timeouts.

[Thomas Kenyon]

Using Asterisk 1.2.12.1.

I have 4 queues running on a server with various handsets logged into them.
When a call comes in, asterisk tries forwarding the call to all 
handsets, including ones that are in use (whereby it gets a BUSY HERE 
response, which is all what you'd expect after all asterisk doesn't know 
how many handsets are on each channel).

If all the handsets are in use, then asterisk will try calling them 
every 30 seconds.

If I have call-limit=1 in sip.conf, will that mean that asterisk will 
try to forward the call as soon as a handset becomes free? (including 
wrapuptime)

If this isn't the case is there anything I can set to allow this?

Should I set call-limit=1 on the peer definitions as a matter of course 
anyway?

I don't want to just look-and-see since this is running on a production 
machine and my test machine doesn't have queues installed and is running 
a completely different version of asterisk.

On a slightly different tone, has anyone written a queue viewer that 
runs as a daemon and serves the pages to the viewer rather than creates 
a manager login/logout event every few seconds? (If not I'll write one 
myself, but worth checking first).