[asterisk-bugs] [Asterisk 0017922]: DTMF not logged to console when configured in logger.conf
Asterisk Bug Tracker
noreply at bugs.digium.com
Sun Aug 29 14:35:10 CDT 2010
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: new
Asterisk Version: 1.6.2.11
JIRA:
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: 2010-08-29 14:35 CDT
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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
======================================================================
----------------------------------------------------------------------
(0126426) jkister (reporter) - 2010-08-29 14:35
https://issues.asterisk.org/view.php?id=17922#c126426
----------------------------------------------------------------------
Confirmed this behavior does not exist sip extension <-> sip extension on
the same pbx.
Issue History
Date Modified Username Field Change
======================================================================
2010-08-29 14:35 jkister Note Added: 0126426
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