[asterisk-r2] Sending DTMF 'f' ?

Amilcar Silvestre amilcar at vonix.com.br
Wed Mar 18 15:47:03 CDT 2009 

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.
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> --Bandwidth and Colocation Provided by http://www.api- 
>>>>> digital.com--
>>>>>
>>>>> asterisk-r2 mailing list
>>>>> To UNSUBSCRIBE or update options visit:
>>>>>   http://lists.digium.com/mailman/listinfo/asterisk-r2
>>>>
>>>>
>>>>
>>>> --
>>>> Douglas Fernando Fischer
>>>> Engº de Controle e Automação
>>>> _______________________________________________
>>>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>>>
>>>> asterisk-r2 mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>>   http://lists.digium.com/mailman/listinfo/asterisk-r2
>>>>
>>>> _______________________________________________
>>>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>>>
>>>> asterisk-r2 mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>>   http://lists.digium.com/mailman/listinfo/asterisk-r2
>>>>
>>>
>>>
>>>
>>> --
>>> "I do not agree with what you have to say, but I’ll defend to the
>>> death your right to say it." Voltaire
>>>
>>> _______________________________________________
>>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>>
>>> asterisk-r2 mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-r2
>>
>>
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-r2 mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-r2
>>
>
>
>
> -- 
> "I do not agree with what you have to say, but I’ll defend to the
> death your right to say it." Voltaire
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-r2 mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-r2

More information about the asterisk-r2 mailing list