On 11/1/06, Leo Ann Boon <[EMAIL PROTECTED]> wrote:
Rajkumar S wrote: > On 10/31/06, Leo Ann Boon <[EMAIL PROTECTED]> wrote: >> Someone correct me if I'm wrong: The Dial string is missing a '/n' >> parameter for the Local channel. Without /n, Asterisk will do a native >> transfer to SIP/1001 and lose the timeout value defined earlier. > > What does '/n' refer here? There is no mention about this in the wiki. > It's in the wiki, see this: http://www.voip-info.org/wiki/index.php?page=Asterisk+local+channels
Thanks Leo. I went though the code of the app_queue to find out if the cutoff value I gave in the dialplan is indeed being passed though when bridging is happening and it's not. The actual line where bridging is happening is bridge = ast_bridge_call(qe->chan,peer, &bridge_config); The bridge_config is of type ast_bridge_config and holds the options to use for this bridging and it has a field called timelimit, which holds the timelimit of the call. This variable is not set in app_queue. This is the reason why the timelimit was not working when called from queue. I edited the code and put a sample value (in milliseconds) and the call cutoff is working fine. I am not sure if this introduces any side effects, but it's so far so good. Another advantage of this method is that the call cutoff will work only when the call is bridged from queue and not from directly called calls. Thanks for your help, Leo and Lenz. raj _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
