I'm interested if anyone else in the Asterisk list can get through to +1-907-747-8633 via voip
Sure, no problem. A nice friendly female voice tells you the time and temp, indeed. The thing is that the call never connects -- that info is sent via call progress, so a misconfigured server (i.e. one that uses the "r" option in dial() or equivalent) would just give you ringing and ringing... [Sep 21 17:49:45] -- Called [EMAIL PROTECTED] [Sep 21 17:49:45] -- SIP/trunks-094da090 is making progress passing it to SIP/1001-b7a030f8 [Sep 21 17:49:48] -- Ringing [Sep 21 17:49:48] -- Progress [Sep 21 17:49:48] -- Peer audio RTP is at port 1.2.3.4:12345 etc. --Luki _______________________________________________ --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
