No subject

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> w=
rote:
> 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=3Dmfcr2
>>> mfcr2_variant=3Dbr
>>> mfcr2_get_ani_first=3Dno
>>> mfcr2_max_ani=3D10
>>> mfcr2_max_dnis=3D4
>>> mfcr2_category=3Dnational_subscriber
>>> mfcr2_logdir=3Dspan1
>>> mfcr2_logging=3Dall
>>> mfcr2_metering_pulse_timeout=3D500
>>> 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
>>> Eng=C2=BA de Controle e Automa=C3=A7=C3=A3o
>>> _______________________________________________
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>>
>>
>>
>> --
>> "I do not agree with what you have to say, but I=E2=80=99ll defend to th=
e
>> death your right to say it." Voltaire
>>
>> _______________________________________________
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>>
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>
>
> _______________________________________________
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>
> asterisk-r2 mailing list
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> =C2=A0 http://lists.digium.com/mailman/listinfo/asterisk-r2
>



--=20
"I do not agree with what you have to say, but I=E2=80=99ll defend to the
death your right to say it." Voltaire