[Freeswitch-users] no DTMF detection over VoIP

[Anita Hall]

If you are using a PRI line, try taking a q931 trace. It will  be able to
show exactly what is happening  below the hood.



regards,
Anita



On Mon, Mar 26, 2012 at 11:48 AM, Neil Patel <neilp at cs.stanford.edu> wrote:

> I guess you do this by using start_dtmf<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf>?
> I tried adding it to my dialplan and also in my lua IVR script:
> session:execute("start_dtmf");
>
> But no luck with either.
>
>
> On Sun, Mar 25, 2012 at 8:40 AM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:
>
>> If the call is coming from a TDM circuit, you'll have to do in-band
>> DTMF detection in that situation.
>>
>> On Sat, Mar 24, 2012 at 6:25 PM, Michael Collins <msc at freeswitch.org>
>> wrote:
>> > Neil,
>> >
>> > Howdy! Long time no chat. :)
>> >
>> > Regarding SIP/VoIP: A SIP call will not come in over a PRI circuit. SIP
>> is
>> > VoIP and PRI is TDM - two totally different methods for carrying
>> telephone
>> > calls. My guess is that you might have a bit of confusion about where
>> and
>> > how calls are coming in. Let's start with the basics: Do you have a
>> SIP/VoIP
>> > provider? If so, did they supply you with phone number(s) for dialing
>> in to
>> > your system? Have you gone through the configuration process to make
>> sure
>> > your FS box is properly communicating with the provider?
>> >
>> > If you need some help you can contact me off list. Of course, this means
>> > you'll have to take me to that Mexican food place the next time I'm in
>> > Frisco! :)
>> >
>> > -MC
>> >
>> >
>> > On Sat, Mar 24, 2012 at 11:18 AM, Neil Patel <neilp at cs.stanford.edu>
>> wrote:
>> >>
>> >> I have not modified the sip_profiles directory from standard. There is
>> no
>> >> variable set for dtmf-type in either internal.xml or external.xml.
>> >>
>> >> I did a trace using:
>> >> sofia global siptrace on
>> >>
>> >> There wer no SIP packets logged. If the SIP call is going to a PRI
>> profile
>> >> (freetdm), will there still be SIP traffic?
>> >>
>> >> Thanks,
>> >> Neil
>> >>
>> >>
>> >> On Sat, Mar 24, 2012 at 7:22 PM, Anita Hall <anita.hall at simmortel.com>
>> >> wrote:
>> >>>
>> >>> What is your config for DTMF in sip profile ?
>> >>>
>> >>> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#dtmf-type
>> >>>
>> >>> Use tshark to generate a SIP trace and paste it on
>> >>> http://pastebin.freeswitch.org/ and give the link here.
>> >>>
>> >>> regards,
>> >>> Anita
>> >>>
>> >>>
>> >>>
>> >>> On Sat, Mar 24, 2012 at 2:43 PM, Neil Patel <neilp at cs.stanford.edu>
>> >>> wrote:
>> >>>>
>> >>>> Hi All,
>> >>>>
>> >>>> I have a basic IVR application in Lua connected to a PRI line. It
>> >>>> currently is not responding to DTMF input given from any VoIP call
>> (e.g.
>> >>>> Skype). However, it accepts input from local mobile or landline
>> calls just
>> >>>> fine.
>> >>>>
>> >>>> I recently pulled latest from git and built; before that, DTMF
>> detection
>> >>>> from both were working. Is there something I need to configure to
>> allow DTMF
>> >>>> for VoIP?
>> >>>>
>> >>>> Thanks,
>> >>>> Neil
>> >>>>
>> >>>>
>> >>>>
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>
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>
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