Jim Boykin pisze:
> We are running Asterisk SVN. We are facing a strange and repetable
> problem. All outgoing call gets terminated in approx 20 minutes.
> Asterisk initiates BYE message to the remote end and call terminates.
>   
Sesion-timer set but not supported by sip-peers?



_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to