On 7/8/10 5:07 AM, "Paul Belanger" <[email protected]> wrote:
> On Wed, Jul 7, 2010 at 2:46 PM, Mike Ely <[email protected]> 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. Got it. The issue was in the "Channel" directive in my AMI script. Before, it looked like this: Action: Originate Channel: SIP/ShoreTel Exten: 7979 Variable: Data=testing1 Context: accept Priority: 1 When it works, it looks like this: Action: Originate Channel: SIP/ShoreTel/7979 Variable: Data=testing1 Context: accept Priority: 1 Thanks for your help! Mike -- _____________________________________________________________________ -- 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
