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

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Sep 9 15:52:33 CDT 2010


A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=17922 
====================================================================== 
Reported By:                jkister
Assigned To:                
====================================================================== 
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:              
====================================================================== 
Date Submitted:             2010-08-27 21:57 CDT
Last Modified:              2010-09-09 15:52 CDT
====================================================================== 
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


======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
has duplicate       0017043 dtmf logging to console not working
====================================================================== 

---------------------------------------------------------------------- 
 (0126791) russell (administrator) - 2010-09-09 15:52
 https://issues.asterisk.org/view.php?id=17922#c126791 
---------------------------------------------------------------------- 
Can you attach a packet capture?  Are you _sure_ the DTMF is going through
Asterisk?  You're only going to see the DTMF log entries if Asterisk
processes the DTMF.  Based on the information we have so far, it's likely
that the media stream (including the DTMF) was directed away from Asterisk
using a re-INVITE. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-09-09 15:52 russell        Note Added: 0126791                          
======================================================================




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