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

[Sherwood McGowan]

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?) 
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....



Also, is there any way you can tell me why asterisk is set to trustrpid and
sendrpid but is still manually taking the callerid info from a user's
configuration instead of honoring my request to
SetCallerPres(prohib_passed_screen) ? I've asked twice on Asterisk-Users and
once on -Dev and haven't gotten any response. I'm about to post it as a bug,
because I think that's what it is...

Thanks again!

Sherwood