[Asterisk-Users] extensions problem (SIP)
Olle E. Johansson
oej at edvina.net
Mon Mar 15 13:56:04 MST 2004
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
> 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
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