[asterisk-r2] Sending DTMF 'f' ?

[Moises Silva]

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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