[asterisk-users] Lastest SVN (1.4) and realtime call limit

Olle E Johansson oej at edvina.net
Thu Feb 22 09:12:07 MST 2007


22 feb 2007 kl. 16.38 skrev Yehavi Bourvine +972-8-9489444:

> Hello,
>
>   I am running version 1.4 with realtime support. I've set (for  
> Snom phones
> 300/320/360) a call limit of 1 (incominglimit and outgoinglimit  
> fields in the
> database).
>
> - When I used 1.4 SIP SHOW PEER show that it has a call limit of 1.  
> The problem
>   was that when such a phone received a call and did attended  
> transfer it
>   was left "in use" and could not receive new calls.
>
> - After seeing reference to similar problem on this list I;ve  
> downloaded today
>   the latest SVN source code and installed it. The problem is that  
> it shows
>   the call limit as 0 and not as 1.
>
> Any idea?

Call limits are in memory flags that we don't keep in the database.
Realtime peers are *not* by default kept in memory and not guaranteed
to stay in memory. Using call limits on them might work, but is not
guaranteed to work.

Realtime peers/users are made to be optimal for large installations,
but lack a lot of the features in regards to call limits, subscriptions,
message waiting indications.

The bug where the transferer was kept in use after the transfer was  
fixed
a few days ago in 1.4 svn.

/O


More information about the asterisk-users mailing list