[asterisk-users] They ignore my DTMF!

[Julio Arruda]

Benjamin Jacob wrote:
> rfc2833 is the prefered way, as inband will work perfectly only with the 
> ulaw codec.
> 

Out of curiosity, there is any 'document' about how RFC2833 would be 
better or worse than SIP INFO ?



> Pierre Marceau wrote:
> 
>> Okay, in the SPA-941 admin I changed:
>>
>> ;DTMF Tx Method: Auto
>> DTMF Tx Method: Inband
>>
>> and now it works.
>>
>> Thanks!
>> Pierre
>>
>>  
>>
>>>>> benjamin.jacob at mgl.com 2/21/2007 12:09 AM >>>
>>>>>       
>> Pierre,
>> Thats exactly what  Joanna  said in her reply.
>> Check the client DTMF settings on your phones.
>> set it to rfc2833 or out-of-band, whatever the config says.
>>
>> Grandstream by default have inband DTMF set, and usualy ulaw is 
>> supported as well, and thats the reason ur grandstream works but 
>> others dont.
>>
>> cheerz
>> - Ben.
>>
>> Pierre Marceau wrote:
>>
>>  
>>
>>> Hi Joanna,
>>>
>>> Thanks for your reply.
>>>
>>> In my mind I think it must be some setting in the client (phone) 
>>> becasue the Grandstream GXP 2000 does work and it is using the same 
>>> sip.conf
>>>
>>> Extensions:
>>> 6000 is xlite softfone
>>> 6003 is Linksys SPA941
>>> 6004 is Grandstream GXP 2000
>>> 6005 is Linksys PAP2NA
>>>
>>> Please have a look at my sip conf and suggest any changes I could try...
>>>
>>> [general]
>>> context=internal
>>> bindport=5060
>>> bindaddr=0.0.0.0
>>> srvlookup=yes
>>> type=friend
>>> secret=XXXXXXX
>>> nat=no
>>> host=dynamic
>>> dtmfmode=rfc2833
>>> disallow=all
>>> allow=ulaw
>>> subscribecontext=internal
>>> canreinvite=no
>>> register=8885551234:XXXXXXXXXXXX at proxy.atlasvoice.com
>>> [atlasvoice]
>>> type=friend
>>> host=proxy.atlasvoice.com
>>> username=8885551234
>>> secret=XXXXXXX
>>> fromuser=8885551234
>>> fromdomain=proxy.atlasvoice.com
>>> canreinvite=no
>>> insecure=very
>>> nat=yes
>>> context=incoming
>>>
>>> [6000]
>>> mailbox=6000 at internal
>>> [6001]
>>> [6003]
>>> [6004]
>>> [6005]
>>> [6006]
>>> [6007]
>>> [6008]
>>>
>>>
>>> Thanks,
>>> Pierre
>>>
>>>
>>>
>>>
>>>   
>>>>>> joannaliza at gmail.com 2/20/2007 10:47 PM >>>
>>>>>>      
>>>>>>         
>>> Hi Pierre,
>>>
>>> Just a thought..check your dtmfmode in your SIP client configuration, if
>>> your using inband but your codec is not ulaw or alaw the DTMF tones 
>>> will be
>>> misrepresented and thus will not be recognised due to the audio 
>>> compression,
>>> on the other hand if your phones are rfc2833 and asterisk is set to 
>>> inband
>>> you wont hear anything.
>>>
>>> Hope that helps.
>>>
>>> Best Regards,
>>> Joanna
>>>
>>> On 2/21/07, Pierre Marceau <pierre at forestcitynetwerxs.com> wrote:
>>>
>>>
>>>   
>>>> Hello,
>>>>
>>>> I can call out to the PSTN and talk to people but when I have to 
>>>> enter a
>>>> dtmf tone in an ivr or voicemail system those systems do not 
>>>> recognise that
>>>> I have sent a tone. This is the case when I make the call with the 
>>>> Xlite
>>>> softfone or a regular telephone plugged into a PAP2NA or a Linksys 
>>>> SPA941.
>>>>
>>>> However... a Grandstream GXP2000 works just great ???
>>>>
>>>> All are extensions on my Asterisk 1.4 box. I am using a voip trunk 
>>>> through
>>>> Atlasvoice. All extensions are setup identical in sip.conf.
>>>>
>>>> One last thing, if a system wants me to respond 1 for sales 2 for 
>>>> service
>>>> I can hit the 1 button quickly 4 or 5 times and the remote system 
>>>> will get
>>>> it. That does not work for a three digit extension as you may well 
>>>> imagine.
>>>>
>>>> Any help would be appreciated.