Re: [asterisk-users] Can't dial out through AMI
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
Re: [asterisk-users] Can't dial out through AMI
On 7/8/10 5:07 AM, Paul Belanger paul.belan...@polybeacon.com wrote: 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. 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
Re: [asterisk-users] Can't dial out through AMI
On 7/6/10 8:44 PM, Mike Ely mike...@amyskitchen.net wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of Paul Belanger Sent: Tue 7/6/2010 5:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject:Re: [asterisk-users] Can't dial out through AMI On Tue, Jul 6, 2010 at 8:00 PM, Mike Ely mike...@amyskitchen.net wrote: Log attached. --- SIP read from UDP:10.10.10.16:5060 --- SIP/2.0 401 Unauthorized context from sip.conf: [ShoreTel] type=peer qualify=yes port=5060 host=10.10.10.16 context=incoming canreinvite=no Your context is not setup properly for outbound, you have no credentials defined. None needed on the ShoreTel side and as I mentioned before regular SIP users can dial out through the Asterisk box using this trunk. Keep in mind, this is a development system on a tightly-controlled network, and I'm trying to start with the simplest case possible, which includes no digest auth for the trunk connection. 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. -- _ -- 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
Re: [asterisk-users] Can't dial out through AMI
On Tue, Jul 6, 2010 at 4:10 PM, Mike Ely mike...@amyskitchen.net wrote: Obviously, I'm playing around with the context a bit but for now just want to get the outbound call working. debug log would be helpful: http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt -- 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
Re: [asterisk-users] Can't dial out through AMI
Log attached. It looks like the call is trying to do an invite to the sip trunk and fails there - it never actually tries to send the destination to the ShoreTel system on the other end of the trunk. Here's the ShoreTel context from sip.conf: [ShoreTel] type=peer qualify=yes port=5060 host=10.10.10.16 context=incoming canreinvite=no On 7/6/10 4:21 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Tue, Jul 6, 2010 at 4:10 PM, Mike Ely mike...@amyskitchen.net wrote: Obviously, I'm playing around with the context a bit but for now just want to get the outbound call working. debug log would be helpful: http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.t xt nosiptrunk.txt Description: Binary data -- _ -- 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
Re: [asterisk-users] Can't dial out through AMI
On Tue, Jul 6, 2010 at 8:00 PM, Mike Ely mike...@amyskitchen.net wrote: Log attached. --- SIP read from UDP:10.10.10.16:5060 --- SIP/2.0 401 Unauthorized context from sip.conf: [ShoreTel] type=peer qualify=yes port=5060 host=10.10.10.16 context=incoming canreinvite=no Your context is not setup properly for outbound, you have no credentials defined. -- 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
Re: [asterisk-users] Can't dial out through AMI
-Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of Paul Belanger Sent: Tue 7/6/2010 5:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject:Re: [asterisk-users] Can't dial out through AMI On Tue, Jul 6, 2010 at 8:00 PM, Mike Ely mike...@amyskitchen.net wrote: Log attached. --- SIP read from UDP:10.10.10.16:5060 --- SIP/2.0 401 Unauthorized context from sip.conf: [ShoreTel] type=peer qualify=yes port=5060 host=10.10.10.16 context=incoming canreinvite=no Your context is not setup properly for outbound, you have no credentials defined. None needed on the ShoreTel side and as I mentioned before regular SIP users can dial out through the Asterisk box using this trunk. Keep in mind, this is a development system on a tightly-controlled network, and I'm trying to start with the simplest case possible, which includes no digest auth for the trunk connection. -- _ -- 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
