[Asterisk-Users] Re: Double DTMF with tdm card
Bart Fisher
asterisk at icpage.com
Thu Nov 3 14:56:30 MST 2005
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" <radamson at routers.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at 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" <asterisk at icpage.com> wrote in message
>> news:005c01c5e0a5$18c22710$f001a8c0 at bart...
>> > OK, then...
>> >
>> > I posted on the Bugs Web Site and markster said: "This is a technical
>> > support issue. Please pursue through Digium tech support
>> > (support at digium.com) 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" <asterisk at linuxguy.com>
>> > To: "Bart Fisher" <asterisk at icpage.com>
>> > Cc: "Walt Reed" <asterisk at linuxguy.com>; "Asterisk Users Mailing List -
>> > Non-Commercial Discussion" <asterisk-users at 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" <asterisk at linuxguy.com>
>> >>> To: "Eric ManxPower Wieling" <eric at fnords.org>
>> >>> Cc: "Walt Reed" <asterisk at linuxguy.com>; "Asterisk Users Mailing
>> >>> List -
>> >>> Non-Commercial Discussion" <asterisk-users at 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" <asterisk at linuxguy.com>
>> >>> >>>>To: <asterisk-users at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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
>
>
More information about the asterisk-users
mailing list