[Asterisk-Users] Re: Double DTMF with tdm card
Bart Fisher
bhfisher at icpage.com
Thu Nov 3 19:20:01 MST 2005
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" <radamson at routers.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at 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 a
> feature
> 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-telephone
> technical
> 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 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 --
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>> >> >>> >>>>>Asterisk-Users mailing list
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>> >> >>> >>>>>
>> >> >>> >>>
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>> > ---------------End of Original Message-----------------
>> >
>> >
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> ---------------End of Original Message-----------------
>
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