[asterisk-bugs] [Asterisk 0017922]: DTMF not logged to console when configured in logger.conf

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17922 
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Reported By:                jkister
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17922
Category:                   Core/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.6.2.11 
JIRA:                       SWP-2134 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-08-27 21:57 CDT
Last Modified:              2010-11-14 00:29 CST
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Summary:                    DTMF not logged to console when configured in
logger.conf
Description: 
duplicate issue of 0017043, except I am using the latest version of the
1.6.2 line of Asterisk compiled from source.

DTMF works correctly; I can see the packets when I tcpdump.  I just never
see any DTMF related information on the console.


pbx1> grep -i dtmf /etc/asterisk/logger.conf 
;    dtmf
console => notice,warning,error,dtmf
pbx1> suex /etc/init.d/asterisk stop
pbx1> ps -ef | grep asterisk
jkister   2359  1598  0 22:50 pts/2    00:00:00 grep asterisk
pbx1> suex /etc/init.d/asterisk start
pbx1> suex asterisk -r
Asterisk 1.6.2.11, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster at digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
details.
This is free software, with components licensed under the GNU General
Public
License version 2 and other licenses; you are welcome to redistribute it
under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.6.2.11 currently running on pbx1 (pid = 2272)
Verbosity is at least 3
pbx1*CLI> core set debug 5
Core debug was 0 and is now 5
pbx1*CLI> core set verbose 5
Verbosity was 3 and is now 5
pbx1*CLI> logger show channels 
Channel                             Type     Status    Configuration
-------                             ----     ------    -------------
/var/log/asterisk/messages          File     Enabled    - Warning Notice
Error 
                                    Console  Enabled    - DTMF Warning
Notice Error 
pbx1*CLI>
pbx1*CLI> 
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
  == Using UDPTL CoS mark 5
    -- Executing [8008374966 at extensions:1] Set("SIP/101-00000006",
"CALLERID(all)=The Kisters <0005551212>") in new stack
    -- Executing [8008374966 at extensions:2] Macro("SIP/101-00000006",
"SaferSIPDial,8008374966") in new stack
    -- Executing [s at macro-SaferSIPDial:1] Set("SIP/101-00000006",
"DIALTRIES=1") in new stack
    -- Executing [s at macro-SaferSIPDial:2] GotoIf("SIP/101-00000006",
"0?unavail") in new stack
    -- Executing [s at macro-SaferSIPDial:3] Set("SIP/101-00000006",
"SIPSERVER=vgw1") in new stack
    -- Executing [s at macro-SaferSIPDial:4] GotoIf("SIP/101-00000006",
"1?preroute") in new stack
    -- Goto (macro-SaferSIPDial,s,10)
    -- Executing [s at macro-SaferSIPDial:10] Set("SIP/101-00000006",
"PREROUTE=9930") in new stack
    -- Executing [s at macro-SaferSIPDial:11] GotoIf("SIP/101-00000006",
"1?dial") in new stack
    -- Goto (macro-SaferSIPDial,s,15)
    -- Executing [s at macro-SaferSIPDial:15] NoOp("SIP/101-00000006",
""Dialing SIP/99308008374966 at vgw1 - try 1"") in new stack
    -- Executing [s at macro-SaferSIPDial:16] Dial("SIP/101-00000006",
"SIP/99308008374966 at vgw1") in new stack
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
  == Using UDPTL CoS mark 5
    -- Called 99308008374966 at vgw1
    -- SIP/vgw1-00000007 is making progress passing it to SIP/101-00000006
    -- SIP/vgw1-00000007 answered SIP/101-00000006
  == Spawn extension (macro-SaferSIPDial, s, 16) exited non-zero on
'SIP/101-00000006' in macro 'SaferSIPDial'
  == Spawn extension (extensions, 8008374966, 2) exited non-zero on
'SIP/101-00000006'
pbx1*CLI>

I entered lots of DTMF going through Verizon's Automated Attendant Hell. 
The DTMF was received by verizon just fine but nothing DTMF related logged
to my console


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Relationships       ID      Summary
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has duplicate       0017043 dtmf logging to console not working
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---------------------------------------------------------------------- 
 (0128843) jkister (reporter) - 2010-11-14 00:29
 https://issues.asterisk.org/view.php?id=17922#c128843 
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same behavior confirmed running Asterisk 1.8.0 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-11-14 00:29 jkister        Note Added: 0128843                          
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