[asterisk-users] They ignore my DTMF!

[Joanna Liza Mariazeta]

Hi Pierre,

You can also add the following if you think its helpful.

relaxdtmf = yes ;relaxes the DTMF detection parameters
qualify = yes  ; will send a SIP Optoin command regularly to check if the
device is still online, if the device did not answer within 2 seconds it
will be considered offline (default time is 2 seconds but can be configured)

Best Regards,
Joanna


On 2/21/07, Benjamin Jacob <benjamin.jacob at mgl.com> wrote:
>
> 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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>
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