[asterisk-r2] Sending DTMF 'f' ?

Moises Silva moises.silva at gmail.com
Wed Mar 18 16:15:22 CDT 2009 

I still fail to see the log where this DTMF comes from chan_dahdi.c

Did you just miss it?

Can you check your log and see if you can find a message like:

DTMF Down 'f'

in chan_dahdi?? otherwise somehow that DTMF is received from somewhere
else, perhaps a manager application or something like that?

On Wed, Mar 18, 2009 at 4:51 PM, Amilcar Silvestre <amilcar at vonix.com.br> wrote:
> Moises,
>
> The log from console, with dtmf debug enabled is:
>
> [Mar 18 17:49:59] DTMF[3411]: channel.c:2279 __ast_read: DTMF begin
> 'f' received on DAHDI/3-1
> [Mar 18 17:49:59] DTMF[3411]: channel.c:2289 __ast_read: DTMF begin
> passthrough 'f' on DAHDI/3-1
> [Mar 18 17:49:59] WARNING[3411]: rtp.c:2205 ast_rtp_senddigit_begin:
> Don't know how to represent 'f'
>
> Amilcar.
>
> On Mar 18, 2009, at 4:47 PM, Amilcar Silvestre wrote:
>
>> Hi Moyses,
>>
>> Ok, I've enabled DTMF debugging, and here's what i've got:
>>
>> [Mar 18 17:34:12] NOTICE[2690] chan_dahdi.c: MFC/R2 call has been
>> accepted on chan 10
>> [Mar 18 17:34:12] NOTICE[2690] chan_dahdi.c: Call accepted on forward
>> channel 10
>> (...)
>> [Mar 18 17:39:51] DTMF[2705] channel.c: DTMF begin 'f' received on
>> DAHDI/10-1
>> [Mar 18 17:39:51] DTMF[2705] channel.c: DTMF begin passthrough 'f' on
>> DAHDI/10-1
>> [Mar 18 17:39:51] WARNING[2705] rtp.c: Don't know how to represent 'f'
>> (...)
>> [Mar 18 17:40:17] DTMF[2705] channel.c: DTMF begin 'f' received on
>> DAHDI/10-1
>> [Mar 18 17:40:17] DTMF[2705] channel.c: DTMF begin passthrough 'f' on
>> DAHDI/10-1
>> [Mar 18 17:40:17] WARNING[2705] rtp.c: Don't know how to represent 'f'
>> (...)
>> [Mar 18 17:40:24] DTMF[2705] channel.c: DTMF begin 'f' received on
>> DAHDI/10-1
>> [Mar 18 17:40:24] DTMF[2705] channel.c: DTMF begin passthrough 'f' on
>> DAHDI/10-1
>> [Mar 18 17:40:24] WARNING[2705] rtp.c: Don't know how to represent 'f'
>> (...)
>> [Mar 18 17:40:44] DTMF[2705] channel.c: DTMF begin 'f' received on
>> DAHDI/10-1
>> [Mar 18 17:40:44] DTMF[2705] channel.c: DTMF begin passthrough 'f' on
>> DAHDI/10-1
>> [Mar 18 17:40:44] WARNING[2705] rtp.c: Don't know how to represent 'f'
>> [Mar 18 17:40:45] DTMF[2793] channel.c: DTMF begin 'f' received on
>> DAHDI/12-1
>> [Mar 18 17:40:45] DTMF[2793] channel.c: DTMF begin passthrough 'f' on
>> DAHDI/12-1
>> [Mar 18 17:40:45] WARNING[2793] rtp.c: Don't know how to represent 'f'
>> (...)
>> [Mar 18 17:43:05] NOTICE[2705] chan_dahdi.c: Chan 10 - Far end
>> disconnected. Reason: Forced Release
>> [Mar 18 17:43:05] NOTICE[2705] chan_dahdi.c: MFC/R2 call disconnected
>> on chan 10
>> [Mar 18 17:43:05] NOTICE[4775] chan_dahdi.c: MFC/R2 call end on chan
>> 10
>>
>> And this is happening in two different telcos.
>>
>> Yes, I've patched asterisk. The patch is that from googlecode for
>> 1.4.23, with a very tiny modification to make the channel returns the
>> hangupcause.
>>
>> Amilcar.
>>
>>
>> On Mar 18, 2009, at 4:32 PM, Moises Silva wrote:
>>
>>> Hum, I see what you mean. But, MFC and DTMF use different pair of
>>> frequencies and F does not even exist in DTMF. Please enable dtmf
>>> debugging in your logger.conf and try to reproduce, I'd expect to see
>>> clearly if chan_dahdi/chan_zap is the one detecting a MF digit and
>>> wrongly sending it down to the core as DTMF digit.
>>>
>>> Did you patched that asterisk yourself?
>>>
>>> From a quick look at the code, the 'f' frame subclass is also used
>>> for
>>> FAX, so that f does not necessarily refers to the MF F tone.
>>>
>>> Moy
>>>
>>> On Wed, Mar 18, 2009 at 4:20 PM, Amilcar Silvestre <amilcar at vonix.com.br
>>>> wrote:
>>>> Hi Moises,
>>>>
>>>> I've already recorded the call using mixmonitor on the sip side. The
>>>> same on the recorded file. Mixmonitor only records the sip side, and
>>>> the far end is muted.
>>>>
>>>> And the problem only occurs on the R2 links. PRI works ok (same
>>>> box).
>>>>
>>>> Seems that, after it enters on the function ast_rtp_senddigit_begin
>>>> in
>>>> rtp.c, the function returns 0 (it doesn't have the "f" digit for
>>>> DTMF,
>>>> the digit 'f' is related to MFC). After it returns 0, the audio from
>>>> DAHDI stops been redirected to the sip end point.
>>>>
>>>> Amilcar.
>>>>
>>>> On Mar 18, 2009, at 4:08 PM, Moises Silva wrote:
>>>>
>>>>> I think you should try to post this in asterisk-users, this is not
>>>>> an R2 issue.
>>>>>
>>>>> Having said that, it would be a good idea to reproduce, and then,
>>>>> when
>>>>> you have a call like that, use dahdi_monitor or ztmonitor to verify
>>>>> the audio is getting into the board correctly, then you can monitor
>>>>> the RTP traffic on the SIP side of the call and see if Asterisk is
>>>>> still sending the audio correctly, if it does, then the problem is
>>>>> definitely not even in Asterisk.
>>>>>
>>>>> On Wed, Mar 18, 2009 at 3:54 PM, Amilcar Silvestre <amilcar at vonix.com.br
>>>>>> wrote:
>>>>>> Douglas,
>>>>>> If you meant the CAS bits, is 1101.
>>>>>> If you mean the configuration of the link:
>>>>>> signalling=mfcr2
>>>>>> mfcr2_variant=br
>>>>>> mfcr2_get_ani_first=no
>>>>>> mfcr2_max_ani=10
>>>>>> mfcr2_max_dnis=4
>>>>>> mfcr2_category=national_subscriber
>>>>>> mfcr2_logdir=span1
>>>>>> mfcr2_logging=all
>>>>>> mfcr2_metering_pulse_timeout=500
>>>>>> Regards,
>>>>>> Amilcar.
>>>>>>
>>>>>> On Mar 18, 2009, at 3:50 PM, Douglas Fischer wrote:
>>>>>>
>>>>>> What do you have in your start bits? (zapata.conf)
>>>>>>
>>>>>> 2009/3/18 Amilcar Silvestre <amilcar at vonix.com.br>
>>>>>>>
>>>>>>> Hi,
>>>>>>>
>>>>>>> I have a box using asterisk 1.4.23.2, and OpenR2 1.1.0. The board
>>>>>>> is a
>>>>>>> Sangoma A104D, From times to times, it shows this message:
>>>>>>>
>>>>>>> WARNING[31548] rtp.c: Don't know how to represent 'f'
>>>>>>>
>>>>>>> Seems to be something related to sending a DTMF digit. After this
>>>>>>> message, the caller (an internal SIP endpoint) doens't hear
>>>>>>> anything
>>>>>>> more from the called (far end, coming from a R2 link), but the
>>>>>>> far
>>>>>>> end
>>>>>>> keeps receiving the audio from the SIP end point.
>>>>>>>
>>>>>>> The message comes with no intervention from the user on SIP end
>>>>>>> point.
>>>>>>>
>>>>>>> Does anyone knows what is happening?
>>>>>>>
>>>>>>> Thanks,
>>>>>>> Amilcar.
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
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>>>>>>
>>>>>>
>>>>>> --
>>>>>> Douglas Fernando Fischer
>>>>>> Engº de Controle e Automação
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>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> "I do not agree with what you have to say, but I’ll defend to the
>>>>> death your right to say it." Voltaire
>>>>>
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>>>
>>>
>>>
>>> --
>>> "I do not agree with what you have to say, but I’ll defend to the
>>> death your right to say it." Voltaire
>>>
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death your right to say it." Voltaire

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