[Freeswitch-users] Use inband dtmf only

[Oleg Stolyar]

Hi Nikolay,

On the advice of the FS team I was able to fix my issue by using spandsp
tone detection instead of the original one.

Basically, in the dialplan use spandsp_start_dtmf instead of start_dtmf.

If you miss some inband DTMF but some are still coming through from that
provider, you might want to give it a try to improve detection.

On Tue, Feb 14, 2017 at 7:46 AM, Oleg Stolyar <olegstolyar at gmail.com> wrote:

> I am having a similar issue with a provider that only supports inband
> DTMF.  About half the time DTMF does not reach FS (or does not register).
>
> I was wondering if enabling jitter buffer on FS might help with this.
> Anyone has experience with it?
>
> On Sun, Feb 12, 2017 at 12:46 AM, Nikolay Zaytsev <nzaytsevc at gmail.com>
> wrote:
>
>> Hello,
>>
>> I have a problem with an inbound provider: sometimes dtmfs do not reach
>> the freeswitch.
>> Provider told me that occurred because I am using G711 with rfc2833, and
>> I should use only G711.
>>
>> Inbound call from the carrier to FreeSwitch:
>>
>> SDP in the INVITE carrier --> FreeSwitch
>> v=0
>> o=Sansay-VSXi 188 1 IN IP4 x.x.x.x
>> s=Session Controller
>> p=+X XXXXXXXX
>> c=IN IP4 x.x.x.x
>> t=0 0
>> m=audio 17774 RTP/AVP 18 0 8 101
>> a=rtpmap:18 G729/8000
>> a=fmtp:18 annexb=no
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=ptime:20
>>
>> SDP in the 200 OK message from FreeSwitch-->carrier:
>>
>> v=0
>> o=FreeSWITCH 1486794789 1486794790 IN IP4 x.x.x.x
>> s=FreeSWITCH
>> c=IN IP4 x.x.x.x
>> t=0 0
>> m=audio 30088 RTP/AVP 0 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>>
>> Carrier wants me to get rid of those session attributes in my 200 OK
>> message:
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> and media to be like this
>> m=audio 30088 RTP/AVP 0.
>>
>> Could you please advise me how to do it?
>>
>> I have already tried to set up
>> dtmf-type=none and liberal-dtmf=false in SIP profiles, dtmf_type=none in
>> Dialplan. In addition I tried start_dtmf before application answer.
>> Also in the vars.xml I specified to use PCMU and PCMA only for both
>> global_codec_prefs and outbound_codec_prefs.
>>
>> My dialplan instruction for inbound call:
>>
>>  <extension name="inbound_call" continue="false">
>>             <condition field="destination_number" expression="^79" >
>>                 <action application="answer"/>
>>                 <action application="sleep" data="1000"/>
>>                 <action application="socket" data="127.0.0.1:8800 async
>> full"/>
>>             </condition>
>>         </extension>
>>
>> Im using several FreeSwitch boxes, one of them is 1.6.2 version another
>> one is 1.6.14, but this problem occurres on both.
>>
>> Thank you for your time and best regards,
>> Nikolay Zaytsev
>>
>> _________________________________________________________________________
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>>
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>>
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>
>
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