[Freeswitch-users] no DTMF detection over VoIP

[curriegrad2004]

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
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
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>>>
>>>
>>> _________________________________________________________________________
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>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
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>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
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>>
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>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
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> http://wiki.freeswitch.org
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>
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