[asterisk-users] DTMF Tones occuring randomly

[Stefan Agethen]

Hi,

I have asked this question months ago - i have "toggled down" all DTMF 
Recognizations in my Asterisk (no more features etc)
and found more people which recognized the same problem, but i cant find 
any help for them and me.

The Problem (short as possible) :

In a randomly call in my business day some unit in my Asterisk System 
sends an randomly DTMF Tone, like "A" "0" or something that do something 
like "#" or "*".
In my case, the "*" let Asterisk hang up my call, i searched for help, 
but nobody knows what to do - so i disabled the "hangup feature" and so 
on, but the problem still exists :(

I sets the hangup-function to :
== Remapping feature Disconnect Call (disconnect) to sequence '*0'

My System is a : Asus with an AMD Athlon XP 3000+ with 512MB of RAM, 1 
Wildcard TDM40B, 2 HFC ISDN PCI Cards from Acer (128k Surf).
Installed is : Debian 3.1 with unstable packages to get Kernel 2.6.15-1 
(AMD Kernel)
(in earlier days my ISDN Driver, mISDN only works with Kernel 2.6.12 or 
higher, Debian is 2.6.8, so...)
The needed Packages for Asterisk are installed (My Installation 
Step-by-Step in german is here : 
http://www.ip-phone-forum.de/showpost.php?p=657963&postcount=7)
Zaptel 1.2.9
Asterisk 1.2.12.1
mISDN in 0.3.0 RC 23
I have changed mpeg123 against madplay.

The Problem exists since a half year or more, i like to say it in 
another way : i have RECOGNIZED the problem since a half year,
i have done many updates of all packages and a clean install to merge 
this prob, no luck, it still exists.

The facts i know about it :

During such a " * DTMF Shooting" the logfiles recognized this (see the 
channel types!) :

-- NOTICES --
Nov  6 09:53:26 WARNING[22637] res_features.c: Bridge failed on channels 
mISDN/1-1 and Zap/1-1
Nov  6 10:05:28 WARNING[22902] res_features.c: Bridge failed on channels 
SIP/40-0815e778 and SIP/pbx1-08281bc8
Nov  6 10:15:38 WARNING[23393] res_features.c: Bridge failed on channels 
SIP/40-0826c530 and IAX2/pbx1-1

DTMF Tone Log :
Nov  6 05:00:33 DTMF[18215] channel.c: SIP/50-0824f1e0 : A
Nov  6 08:44:05 DTMF[21660] channel.c: Zap/1-1 : A
Nov  6 09:43:00 DTMF[22520] channel.c: SIP/pbx1-08274fb8 : 0
Nov  6 09:53:26 DTMF[22637] channel.c: Zap/1-1 : *
Nov  6 10:05:28 DTMF[22902] channel.c: SIP/pbx1-08281bc8 : *
Nov  6 10:14:42 DTMF[23288] channel.c: mISDN/2-1 : 8
Nov  6 10:16:11 DTMF[23426] channel.c: SIP/pbx1-08274690 : *
Nov  6 10:17:45 DTMF[23288] channel.c: Zap/1-1 : A
Nov  6 10:32:54 DTMF[23545] channel.c: Zap/1-1 : D
Nov  6 10:35:58 DTMF[23792] channel.c: SIP/pbx1-08273ef8 : *

-- ASTERISK SIP DEBUG (one case) --
Nov  6 10:35:54 DEBUG[23792] channel.c: Got DTMF on channel 
(SIP/40-0825b3c8)
Nov  6 10:35:54 DEBUG[23792] channel.c: Bridge stops bridging channels 
SIP/40-0825b3c8 and SIP/pbx1-08273ef8
Nov  6 10:35:54 DEBUG[23792] res_features.c: Feature interpret: 
chan=SIP/40-0825b3c8, peer=SIP/pbx1-08273ef8, sense=1, features=18
Nov  6 10:35:54 DEBUG[23792] res_features.c: Set time limit to 500     
Nov  6 10:35:55 DEBUG[23792] channel.c: Nobody there, continuing...      
Nov  6 10:35:58 DEBUG[23792] channel.c: Bridge stops because we're 
zombie or need a soft hangup: c0=SIP/40-0825b3c8, c1=SIP/pbx1-08273ef8, 
flags: No,Yes,No,No
Nov  6 10:35:58 DEBUG[23792] channel.c: Bridge stops bridging channels 
SIP/40-0825b3c8 and SIP/pbx1-08273ef8
Nov  6 10:35:58 DEBUG[23792] res_features.c: Timed out for feature!        
Nov  6 10:35:58 DEBUG[23792] channel.c: Hanging up channel 
'SIP/pbx1-08273ef8'
Nov  6 10:35:58 DEBUG[23792] chan_sip.c: Hangup call SIP/pbx1-08273ef8, 
SIP callid 51fe564f078ca6db08c1edb1367ca701 at pbx-network.de)
Nov  6 10:35:58 DEBUG[23792] chan_sip.c: 
update_call_counter(02088480499) - decrement call limit counter
Nov  6 10:35:58 DEBUG[23792] app_dial.c: Exiting with DIALSTATUS=ANSWER.
Nov  6 10:35:58 DEBUG[23792] pbx.c: Spawn extension (voip_wahl,_X.,6) 
exited non-zero on 'SIP/40-0825b3c8'


As you can see in the DTMF Log - there are many "Digits" send, but they 
dont scare me, but the "*" are disconnecting my calls - thats a problem 
for me and my business..

I HOPE !!! you can help me, Best wishes, Stefan