[asterisk-bugs] [Asterisk 0013254]: Inband dtmf not working in voicemailmain

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Sep 2 14:52:59 CDT 2008


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=13254 
====================================================================== 
Reported By:                ruchirb
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   13254
Category:                   Applications/app_voicemail
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.16.2 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             2008-08-07 11:15 CDT
Last Modified:              2008-09-02 14:52 CDT
====================================================================== 
Summary:                    Inband dtmf not working in voicemailmain
Description: 
OS: Centos 5.0
Asterisk Version : 1.4.16.2
Configuration: Realtime
Description:
When we configure dtmfmode=auto for sip user in database, dtmf mode inband
in phone(linksys pap2) and dial an extension going to voicemailmain, it
doesn't detect dtmf at all. 
If we set dtmfmode=inband then it works. 
I think if we have defined dtmfmode=auto then asterisk should accept
whatever the phone sends, isn't it?
====================================================================== 

---------------------------------------------------------------------- 
 (0091985) blitzrage (administrator) - 2008-09-02 14:52
 http://bugs.digium.com/view.php?id=13254#c91985 
---------------------------------------------------------------------- 
Can you please provide the sip trace along with the console debug output
when you setup the call? Be sure to setup the 'debug' option in logger.conf
for the console and 'logger reload' before outputting the debug information
to a file.

Here is how I get console output:

asterisk -cvvvvvdddddn | tee /tmp/console_output.txt 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-09-02 14:52 blitzrage      Note Added: 0091985                          
======================================================================




More information about the asterisk-bugs mailing list