SIP user => Asterisk 1.6 server => SIP Trunk => external destination:
works
AMI script => Asterisk 1.6 server => SIP Trunk => external destination:
Failed to authenticate on INVITE to '"asterisk"
<sip:asterisk@(ipaddr)>;tag=alphanumeric'
I?ve tried doing things like ?include => contextwithtrunk" in various
places, googling, re-reading relevant portions of the largish O'Reilly
Asterisk book, no avail.
The call will go through to a registered SIP user just fine, but won't seem
to go out off the trunk.
Here's the basic set of commands being sent through AMI:
Action: Originate
Channel: SIP/ShoreTel
Variable: Data=teletubbie-murder
Context: accept
priority: 1
Number: (external number reachable from regular SIP user account)
Here's the accept context:
[accept]
include => incoming
include => outbound-pbx
exten => s,1,Answer
exten => s,n,Playback(custom/msg1)
exten => s,n,Background(custom/how-to-ack)
exten => s,n,WaitExten(5,m)
exten => 1,1,ForkCDR(v,s(fullcmd=${Data}))
exten => 1,n,Background(${Data})
exten => 1,n,Background(discon-or-out-of-service)
exten => 1,n,WaitExten(5,m)
exten => 1,n,Hangup
exten => 2,1,Background(de-activated)
exten => 2,n,ForkCDR(v,s(reject=${Data}))
exten => 2,n,Hangup
exten => 3,1,Goto(accept,1,2)
exten => *,1,Goto(accept,s,1)
exten => i,1,Goto(accept,s,1)
exten => t,1,Goto(accept,s,1)
Obviously, I'm playing around with the context a bit but for now just want
to get the outbound call working.
On Tue, Jul 6, 2010 at 4:10 PM, Mike Ely <mikeely at 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.belanger at polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com
-----Original Message----- From: asterisk-users-bounces at 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 <mikeely at 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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100706/8e97a6aa/attachment.htm