This is exactly what is happening... It's bad news... In my case the T1 is 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?
>>> >>>>>_______________________________________________
>>> >>>>>--Bandwidth and Colocation sponsored by Easynews.com --
>>> >>>>>
>>> >>>>>Asterisk-Users mailing list
>>> >>>>>Asterisk-Users@lists.digium.com
>>> >>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>> >>>>>To UNSUBSCRIBE or update options visit:
>>> >>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>> >>>>>
>>> >>>>>
>>> >>>
>>> >>>_______________________________________________
>>> >>>--Bandwidth and Colocation sponsored by Easynews.com --
>>> >>>
>>> >>>Asterisk-Users mailing list
>>> >>>Asterisk-Users@lists.digium.com
>>> >>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>> >>>To UNSUBSCRIBE or update options visit:
>>> >>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>> >>>
>>> >>
>>> >_______________________________________________
>>> >--Bandwidth and Colocation sponsored by Easynews.com --
>>> >
>>> >Asterisk-Users mailing list
>>> >Asterisk-Users@lists.digium.com
>>> >http://lists.digium.com/mailman/listinfo/asterisk-users
>>> >To UNSUBSCRIBE or update options visit:
>>> >  http://lists.digium.com/mailman/listinfo/asterisk-users
>>> >
>>> >
>>>
>> _______________________________________________
>> --Bandwidth and Colocation sponsored by Easynews.com --
>>
>> Asterisk-Users mailing list
>> Asterisk-Users@lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>
> _______________________________________________
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



_______________________________________________
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


---------------End of Original Message-----------------


_______________________________________________
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users



_______________________________________________
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to