asterisk users - Nov 2006 - DTMF Tones occuring randomly

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@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

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@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

Stefan Agethen wrote:
> 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 is called Talk Off (or maybe Talkoff).  Search the archives 
for that.

>> Stefan Agethen wrote:
>>/ 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 is called Talk Off (or maybe Talkoff).  Search the archives 
> for that.
Yes, i know, talkoff means that at example a human voice is interpreted as a
DTMF Signal, i have searched for that,
the only possible thing i?ve found is the statement to turn on
"dtmfthreshold" - but this only worked versions ago in zapata.conf, i
think.

I cant find anymore Information to dtmfthreshold.

No more helpful results any more for "my talkoff"... :(

Hi Eric,

i have replied but nobody seems to got a deeper knowledge of the problem.

I have searched for talkoff, i found a lot of stuff, like check IRQs 
(checked, and good) and/or set relaxdtmf=no (it is set)
or check the dtmf modes to be the same or or.

But nothing of the things i found match to my problem except one thing i 
cant understand - there is an thread at digium with the advice to use 
the variable
"dtmfthreshold" to set the level of dtmf detection, i cant find any 
variable like this.

Do you know something where i can search ?

I got this problem since 6 or 7 months and tried MANY solutions to get 
to my stable Asterisk, but i got no luck.

What do you think about switching from rfc2833 to inband to solve this 
problem ?

Thanks, Stefan

Stefan Agethen wrote:
> Hi Eric,
> 
> i have replied but nobody seems to got a deeper knowledge of the problem.
> 
> I have searched for talkoff, i found a lot of stuff, like check IRQs 
> (checked, and good) and/or set relaxdtmf=no (it is set)
> or check the dtmf modes to be the same or or.
> 
> But nothing of the things i found match to my problem except one thing i 
> cant understand - there is an thread at digium with the advice to use 
> the variable
> "dtmfthreshold" to set the level of dtmf detection, i cant find
any
> variable like this.
> 
> Do you know something where i can search ?
> 
> I got this problem since 6 or 7 months and tried MANY solutions to get 
> to my stable Asterisk, but i got no luck.
> 
> What do you think about switching from rfc2833 to inband to solve this 
> problem ?
What codec are you currently using for voice?

I have found that when nothing else works, playing with the gains on the 
Zap channel helped.  Usually lowering them.

> What codec are you currently using for voice?
> I have found that when nothing else works, playing with the gains on the 
> Zap channel helped.  Usually lowering them.
I use rfc2833 for dtmf, alaw as codec.

Yes, a lowering could be a idea, but the problem is logged on any kind of
channels in my system, like zap, misdn, sip and iax.

That is my problem :(

> What codec are you currently using for voice?
> I have found that when nothing else works, playing with the gains on the 
> Zap channel helped.  Usually lowering them.
I use rfc2833 for dtmf, alaw as codec.

Yes, a lowering could be a idea, but the problem is logged on any kind 
of channels in my system, like zap, misdn, sip and iax.

That is my problem :(