[asterisk-bugs] [Asterisk 0010412]: Transfer and One-Touch Recording not working

noreply at bugs.digium.com noreply at bugs.digium.com
Thu Aug 9 01:10:56 CDT 2007


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=10412 
====================================================================== 
Reported By:                cyllene
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   10412
Category:                   Resources/res_features
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.9  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             08-08-2007 19:04 CDT
Last Modified:              08-09-2007 01:10 CDT
====================================================================== 
Summary:                    Transfer and One-Touch Recording not working
Description: 
I am able to establish a call, but issuing the "*1" and "*2" commands do
nothing. There is no output in the console with very high verbosity.

extensions.conf:
exten => 82,1,Set(DYNAMIC_FEATURES=automon)
exten => 82,n,Dial(IAX2/tom,30,HWT)

As you can see, I have W and T enabled. I am trying *1 and *2 from the
calling party (not callee).

features.conf:
[featuremap]
automon => *1
atxfer => *2

The res_features.so module is indeed loaded. Asterisk version is 1.4.10
(not 1.4.9. There is no 1.4.10 in the list)
====================================================================== 

---------------------------------------------------------------------- 
 russell - 08-09-07 01:10  
---------------------------------------------------------------------- 
What kind of phone are you using to call this extension?  If it is SIP,
what dtmf mode are you using?

Also, enable "dtmf" logging in /etc/asterisk/logger.conf for the console. 
That will let you get some debug output from the Asterisk core as it
handles DTMF. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
08-09-07 01:10  russell        Note Added: 0068642                          
======================================================================




More information about the asterisk-bugs mailing list