Re: [Asterisk-Users] Re: Double DTMF with tdm card

Bart Fisher Thu, 03 Nov 2005 18:20:47 -0800

I just heard back from Mark.  I volunteered my system to used for testing.

From Mark:

"Generally, issues which involve Digium hardware should go through
technical support, even if it's a newly introduced problem, because they
can help narrow down the nature of the failure, what might have changed,
etc.  If you or a representative of this "group" want to fill this role
instead, I'm happy to work with you, but I need the situation labbed up in
an environment where the problem can be demonstrated, where I can remotely
log in, and where I can edit, recompile, and test in real time (i.e. not
on a production server).  If you want to set all this up and contact me
with login details and a number where I can see the problem occur, then
when it's ready, I can work with you directly.

Mark"


Bart

----- Original Message -----From: "Rich Adamson" <[EMAIL PROTECTED]>To: "Asterisk Users Mailing List - Non-Commercial Discussion"<asterisk-users@lists.digium.com>

Sent: Thursday, November 03, 2005 2:41 PM
Subject: Re: [Asterisk-Users] Re: Double DTMF with tdm card

"If" in fact it is the exact same issue, then I'd suggest creating afeature

request to add "disable dtmf detection after answer supervision" and post
it to the -dev list (which is what Kevin is suggesting now). You will need

to be explain the wanted functionality in terms that non-telephonetechnical

folks can understand. I'd suggest a zapata.conf configuration option that
is something like "ignore-dtmf-after-answersup" with a default value of
however it works today (=no).

Think about that carefully as the option set to =yes will disable dtmf
from interacting with your internal * ivr (assuming you have one).
What you want is kind of related to a pass-thru connection and not
necessarily for a connection terminating within *. There might be other
ways to handle your objective.

This same issue comes up in other cases where interaction with an external
ivr is needed, some airlines automated systems, etc.

I honestly believe the exact same thing should apply to iax2 incoming
trunks as well. Not so sure about sip trunks.

I'd agree with your statement relative to digium support being contacted,
but if the boss-man suggests it, there might be an unstated reason for
that. If properly worded (and with the supporting documentation that you
heard the problem with a T1 analyzer), they might be able to help support
the need for some kind of option.

------------------------

This is exactly what is happening... It's bad news... In my case the T1is

connected to a PBX Voice Mail.  So, double dialing really messes up thing
like when entering a passcode.  Where passcode "1234" arrives as
"11223344" - no good.   This would always be an issue in cases where the
call is Tandem thru Asterisk.

In fact, I can't see any reason to repeat the digits when the signal is
"inband" and/or Zap Bridged call. -  And why was it changed from 1.0.9?
Makes no sense.

It seems an easy fix, maybe a digit time-out parameter or disable sending
after answer supervision has been achieved.

Given what you say, Digium Support won't be able to fix without code
changes - I don't know what Mark is thinking here.

Bart

----- Original Message -----From: "Rich Adamson" <[EMAIL PROTECTED]>

To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Sent: Thursday, November 03, 2005 1:17 PM
Subject: Re: [Asterisk-Users] Re: Double DTMF with tdm card


>I might be able to shed a little light on this...
>
> Asterisk is constantly listening for dtmf tones on most channels. Its
> either listening for inband or rfc-out-of-band, depending upon how the
> attached device is defined and how asterisk def's for that device is

> defined. For pstn interfaces, the "cards" don't listen for any dtmf,> but

> rather the zap sutff is listening.
>
> If a call is generated from some external source (coming into *), the

> dtmf will be inband once a channel is answered. For commercial> telephone> equipment, once a channel is answered, the telephone equipment no> longer

> listens for dtmf (its simply passed inband). Not so with asterisk, and
> this point has been argued with Mark some time ago; asterisk still

> listens and trys to handle the dtmf, translating to rfc2833 as it> thinks

> is necessary.
>
> So, it sounds like you have an answered T1 call where * is still trying
> to handle dtmf (regenerating it), AND, the dtmf is being passsed inband

> as well. If that is what you are seeing, then its the same design> problem

> that was argued with Mark, and he's insistent the current operation is
> correct. I disagree, but I'm only one person.
>
> ------------------------
>

>> SO is he definitively saying that the asterisk software is not>> involved

>> here? (listening or regenerating tones)
>>

>> -->> -->> Steven

>>

>> May you have the peace and freedom that come from abandoning all hope>> of

>> having a better past.
>> ---    -      ---  - - -       -    -     -   -   --  - - - --- - ------
>>   -
>>  - --- - - -- -  -    - --   -   -    -
>> "Bart Fisher" <[EMAIL PROTECTED]> wrote in message
>> news:[EMAIL PROTECTED]
>> > OK, then...
>> >

>> > I posted on the Bugs Web Site and markster said: "This is a>> > technical

>> > support issue. Please pursue through Digium tech support

>> > ([EMAIL PROTECTED]) and contact me if you have any issues.",>> > Hmmm...

>> >
>> > So I have written support - still waiting for answer - If I hear
>> > anything
>> > I'll let you know....
>> >
>> > Bart
>> >

>> > ----- Original Message ----->> > From: "Walt Reed" <[EMAIL PROTECTED]>

>> > To: "Bart Fisher" <[EMAIL PROTECTED]>

>> > Cc: "Walt Reed" <[EMAIL PROTECTED]>; "Asterisk Users Mailing>> > List -

>> > Non-Commercial Discussion" <asterisk-users@lists.digium.com>
>> > Sent: Thursday, November 03, 2005 9:57 AM
>> > Subject: Re: [Asterisk-Users] Double DTMF with tdm card
>> >
>> >

>> >> Frankly, I think this may be happening to me too. It's still a "zap>> >> to

>> >> zap" channel problem.
>> >>
>> >> On Thu, Nov 03, 2005 at 08:27:59AM -0800, Bart Fisher said:
>> >>> My problem is slightly different as there is 2 T1 Ports involved -
>> >>> With
>> >>> a
>> >>> T1 test set I can clearly hear two tones sent quickly with each
>> >>> outside

>> >>> caller press. I assume one of the tones is the actual audio>> >>> passing

>> >>> thru

>> >>> the connection and the other generated by the T1 card itself.>> >>> If I

>> >>> make

>> >>> the same test with a TDM400 as input connection and the TE410P>> >>> port

>> >>> as
>> >>> output connection, there is no double dialing. Same results if an
>> >>> inside

>> >>> extension is used as input connection. It only happens if it's a>> >>> T1

>> >>> to
>> >>> T1
>> >>> Bridge...
>> >>>

>> >>> If it is a regenerated tone from the TE410, it seems there should>> >>> be

>> >>> some
>> >>> option to stop listening for tone touch after connection has been
>> >>> established?
>> >>>
>> >>> Bart
>> >>>
>> >>>

>> >>> ----- Original Message ----->> >>> From: "Walt Reed" <[EMAIL PROTECTED]>

>> >>> To: "Eric ManxPower Wieling" <[EMAIL PROTECTED]>
>> >>> Cc: "Walt Reed" <[EMAIL PROTECTED]>; "Asterisk Users Mailing
>> >>> List -
>> >>> Non-Commercial Discussion" <asterisk-users@lists.digium.com>
>> >>> Sent: Thursday, November 03, 2005 6:50 AM
>> >>> Subject: Re: [Asterisk-Users] Double DTMF with tdm card
>> >>>
>> >>>
>> >>> >Note this is on external calls to external applications.... Not
>> >>> >Asterisk

>> >>> >DTMF detection. It's as though DTMF is distorted when going>> >>> >through

>> >>> >a
>> >>> >TDM fxs port, or that it's being caught (too late) and then
>> >>> >retransmitted. Does * intercept outgoing dtmf?
>> >>> >
>> >>> >I haven't found good docs that tell exactly what relaxdtmf does.
>> >>> >
>> >>> >On Thu, Nov 03, 2005 at 08:01:03AM -0600, Eric ManxPower Wieling
>> >>> >said:
>> >>> >>Did you try relaxdtmf=no
>> >>> >>
>> >>> >>Walt Reed wrote:

>> >>> >>>Nope - I saw your posts on it though. Very frustrating. I've>> >>> >>>had

>> >>> >>>to

>> >>> >>>discontinue use of my TDM FXS ports until some solution is>> >>> >>>found.

>> >>> >>>
>> >>> >>>On Wed, Nov 02, 2005 at 10:18:47AM -0800, Bart Fisher said:
>> >>> >>>
>> >>> >>>>Did you ever find a solution for this problem?  I have it on
>> >>> >>>>latest
>> >>> >>>>Beta 2
>> >>> >>>>
>> >>> >>>>Bart
>> >>> >>>>
>> >>> >>>>

>> >>> >>>>----- Original Message ----->> >>> >>>>From: "Walt Reed" <[EMAIL PROTECTED]>

>> >>> >>>>To: <asterisk-users@lists.digium.com>
>> >>> >>>>Sent: Friday, October 21, 2005 7:26 AM
>> >>> >>>>Subject: [Asterisk-Users] Double DTMF with tdm card
>> >>> >>>>
>> >>> >>>>
>> >>> >>>>
>> >>> >>>>>I have a TDM22B (latest rev), Sipura 2000, and Cisco ATA 186.
>> >>> >>>>>Running
>> >>> >>>>>CVS HEAD from about a week ago.
>> >>> >>>>>

>> >>> >>>>>Calls made from a SIP device on either the cisco or sipura>> >>> >>>>>work

>> >>> >>>>>fine.
>> >>> >>>>>

>> >>> >>>>>Call made from an analog phone hooked up to one of the FXS>> >>> >>>>>ports

>> >>> >>>>>on
>> >>> >>>>>the
>> >>> >>>>>TDM22B  sound fine, but any DTMF entered after the call is
>> >>> >>>>>bridged
>> >>> >>>>>to
>> >>> >>>>>an
>> >>> >>>>>outside number (like entering a PIN for a bank or external
>> >>> >>>>>conference

>> >>> >>>>>bridge) is frequently doubled. Entering 1234 may be>> >>> >>>>>recognized

>> >>> >>>>>as
>> >>> >>>>>112344 for example.
>> >>> >>>>>

>> >>> >>>>>I ran fxotune, and played with the rx and tx gains a little,>> >>> >>>>>but

>> >>> >>>>>have
>> >>> >>>>>been unable to resolve the problem...
>> >>> >>>>>

>> >>> >>>>>* has no problem dialing outside numbers. It's just DTMf>> >>> >>>>>after

>> >>> >>>>>the
>> >>> >>>>>call
>> >>> >>>>>is bridged between zap channels...
>> >>> >>>>>
>> >>> >>>>>Any ideas?
>> >>> >>>>>_______________________________________________
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Re: [Asterisk-Users] Re: Double DTMF with tdm card

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