[Asterisk-Users] extensions problem (SIP)

[Walker Haddock]

On Mon, Mar 15, 2004 at 09:56:04PM +0100, Olle E. Johansson wrote:
> 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.

See bug (closed),
http://bugs.digium.com/bug_view_page.php?bug_id=0000408

This is the one that Paul Liew opened and provide a fix for.

> 
> 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.

Yes, I have my SIP UAs defined as `friends`.  Also, like Paul said in bug 1064, you have to put the username=xxx in the stanza as well.

> 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
I see that Paul is commenting on this, so I don't need to get involved except to say that I am using the feature to limit the number of calls that * can make to the UA.  If there is a disagreement in definition/perspective here, I guess we can change all of our sip.conf entries from incoming to outgoing easily enough.

Walker
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