[asterisk-bugs] [Asterisk 0013565]: Calls originated from AMI do not have channel variables specified in sip.conf set

Asterisk Bug Tracker noreply at bugs.digium.com
Wed Dec 10 14:13:45 CST 2008


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=13565 
====================================================================== 
Reported By:                ajohnson
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   13565
Category:                   Core/ManagerInterface
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.4.21.2 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             2008-09-26 12:35 CDT
Last Modified:              2008-12-10 14:13 CST
====================================================================== 
Summary:                    Calls originated from AMI do not have channel
variables specified in sip.conf set
Description: 
Console output:
Executing [s at macro-zap-out:2] NoOp("SIP/nottiano-0845a458", "User_ID: ")
in new stack

extensions.conf
exten => s,n,NoOp(User_ID: ${User_ID})

sip.conf
setvar=User_ID=nottiano

Calls originated through the AMI does not have the variable set.  Calls
originated from the SIP peer do have the variable set.
====================================================================== 

---------------------------------------------------------------------- 
 (0096133) file (administrator) - 2008-12-10 14:13
 http://bugs.digium.com/view.php?id=13565#c96133 
---------------------------------------------------------------------- 
As blitzrage mentioned the setvar option is for calls *into* Asterisk. I
have, however, attached a patch that will give you the behavior you seek. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-12-10 14:13 file           Note Added: 0096133                          
======================================================================




More information about the asterisk-bugs mailing list