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

2007-02-22 Thread 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?

 Thanks, __yehavi:
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Re: [asterisk-users] Lastest SVN (1.4) and realtime call limit

2007-02-22 Thread Olle E Johansson


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
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