[asterisk-users] They ignore my DTMF!

[Benjamin Jacob]

rfc2833 is the prefered way, as inband will work perfectly only with the 
ulaw codec.

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.
>>>
>>>Pierre
>>>
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>>>
>>>      
>>>
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>>
>>    
>>
>
>  
>

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