[asterisk-users] They ignore my DTMF!

[Benjamin Jacob]

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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