On Wed, Jul 7, 2010 at 2:46 PM, Mike Ely <mike...@amyskitchen.net> wrote: > Maybe I missed something here? SIP users configured within Asterisk can > dial out just fine through the trunk. It's just when I try to use AMI that > it fails. > The far end is rejecting your call; SIP/2.0 401 Unauthorized.
If you can dialout without using AGI, then capture a 2nd debug log, and post it. We can then compare why one works and the other does not. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
