[Asterisk-Dev] Attempting to make new documentation for newsettings

[Olle E. Johansson]

Sherwood McGowan wrote:
> Olle,
> 
> Thanks for your input. I'll definitely be just adding a section in the
> sip.conf page that says "** New In 1.2+ **". The problem with incominglimit
> and outgoing limit (I haven't changed over to call-limit, will be today)
> causing registration problems is this: 
> 
> Customer's remote asterisk server has registered with mine and is making and
> receiving calls. I have him set to 2 incoming and 2 outgoing channels. (btw,
> would that be a call-limit of 2 or 4?) 
Remember that the user and peer have separate call limits, there is no
direction, just a total call limit. Setting 2 for a type=friend means 2
for the user, 2 for the peer, a total of 4.

> Customer either dials a 3rd call or receives a 3rd call for whatever reason.
> 
> My server then has been disabling making or receiving calls (issuing a sip
> response code I don't remember, but it's Call Limit Reached), and (from the
> customer's words, since I don't see ANYTHING on my server) my system will
> not release that lock until the customer has either done a sip reload or
> until I manually prune the customer's peer/user infos from the realtime
> cache (using sip prune ${NUM}).
> 
> Any thoughts or suggestions as to why this is happening? It makes sense to
> disallow calls until the channels in use have gone back down, but the
> customer says it literally is staying locked out until he sip reloads....
Is this with latest CVS head? Sounds like we do not release the call
limit when he hangs up. Try logging and see if you can find any log
output or missing log output at hangup of the call. Set debug to 4 and
verbose to 4.

Haven't played around at all with RPID stuff, so someone else needs to
answer that part.

/O