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@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@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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