[Asterisk-Users] extensions problem (SIP)

[Olle E. Johansson]

Walker Haddock wrote:
> On Mon, Mar 15, 2004 at 09:28:00PM +0100, Olle E. Johansson wrote:
> 
>>The incominglimit limits how many simultaneous calls a UA may place to 
>>Asterisk.
> 
> I'm pretty sure that the incominglimit specifies how many calls that * can send to the SIP device.  If you set incominglimit=1 and then do a SIP SHOW INUSE from the *CLI then you will see the limit set.  The behavior of * then will consider the device busy if there is a call in progress and the inuse count is incremented.
> 
> Paul Lieu did some work on this a few months ago and I've been using it on my Cisco 7960 and Grandstream BT-102 phones.

The incominglimit= config is read in build_user, users place calls to asterisk.
Peers have no incominglimit.

Funny enough outgoinglimit= was also coded in build_user for users, even though
chan_sip place calls to peers.

So maybe it just happens to work as you say for "friends" that is both user and peer.
The find_user routine just checks users, not peers.

Would be greatful if someone cleaned up this part of chan_sip and added support
for outgoinglimit for peers.

Also, see bug
http://bugs.digium.com/bug_view_page.php?bug_id=0001064

/O